| 00:00.08 | *** join/#asterisk linlin (n=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 00:02.45 | Grubs | Does the * server need a sound card to be able to play music on hold to callers? |
| 00:02.54 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com) |
| 00:03.40 | Ariel_ | Grubs, no |
| 00:03.42 | Grubs | I've tried mpg123 and format_mp3 and with both MOH stops immediately after trying to start. (asterisk 1.2.2) |
| 00:03.58 | Grubs | thx Ariel_ |
| 00:04.04 | Nugget | Maybe asterisk just doesn't like your taste in music. |
| 00:04.22 | Grubs | lol - using the .mp3s that come with * :) |
| 00:05.56 | s34n | I just switched trunking, and extension dialing stopped working |
| 00:06.24 | s34n | I have a WaitExten(), but * doesn't seem to recognize key presses |
| 00:07.03 | s34n | It worked fine this morning with a PRI trunk, but stopped working on the new SIP trunk |
| 00:07.23 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 00:08.47 | Ariel_ | s34n, do you have the correct dtmfmode setting for the sip provider? |
| 00:09.49 | *** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com) |
| 00:10.29 | s34n | Ariel_: good question |
| 00:11.45 | Grubs | Yippee!! just got MP3Player working! next stop MOH. |
| 00:12.03 | Grubs | problem was from running * as non root. |
| 00:12.03 | s34n | anybody know what dtmfmode should be for a metaswitch? |
| 00:13.27 | s34n | Ariel_: got it. thanks. |
| 00:13.43 | s34n | for anyone else who cares, it is inband |
| 00:13.54 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 00:14.14 | Ariel_ | s34n, I was going to say if your using ulaw, you should be using inband. |
| 00:14.30 | Ariel_ | if your using any other then use dtmfmode=rfc2833 |
| 00:15.10 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 00:15.28 | [TK]D-Fender | Ariel_ : Why would you ever want DTMF inband if you could have it OOB? |
| 00:15.56 | [TK]D-Fender | Ariel_ : Adds DSP load on your * side and the possiblity of integrity issues... |
| 00:15.58 | Ariel_ | [TK]D-Fender, lots of ulaw providers only use inband not OOB. |
| 00:16.36 | [TK]D-Fender | Ariel_ : So its not so much a suggest as it is a statisctical fact of whats available then? |
| 00:16.42 | *** join/#asterisk _Angela (n=santiago@208.195.215.222) |
| 00:17.01 | De_Mon | is that true for iax aswell as sip? |
| 00:17.51 | Ariel_ | yes |
| 00:18.03 | Ariel_ | [TK]D-Fender, just know this from working on many asterisk setups. |
| 00:18.30 | De_Mon | I setup teliax and voipjet today and neither specified which dtmf mode to use |
| 00:18.59 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 00:19.18 | [TK]D-Fender | Ariel_ : I've seen a few that have used it before... it irks me... |
| 00:19.52 | Ariel_ | [TK]D-Fender, sorry to hear it. But that is just the fact's.... |
| 00:20.04 | Ariel_ | De_Mon, it's dependent on the codec your using |
| 00:20.35 | De_Mon | so asterisk will know which dtmf mode to use based on the codec and I shouldn't have to specify? |
| 00:20.40 | [TK]D-Fender | Ariel_ : Nothing that actually affect MY life mind you :) I'm on analog w/ SPA-3000 that does DTMF as INFO, and a PRI w/ * that sends it as RFC2833 :) |
| 00:21.25 | [TK]D-Fender | De_Mon : No, Ariel_ was just stating a common implementation. Always check with your provider. |
| 00:21.25 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 00:21.34 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 00:21.43 | Ariel_ | funny pri uses dchannel not rfc2833 |
| 00:22.15 | Ariel_ | [TK]D-Fender, and most of the sipura's I have use inband if it's ulaw other codec they use auto mode which changes to rfc2833. |
| 00:22.31 | [TK]D-Fender | Ariel_ : I'm talking about my PRI at work. I route my home to a work DID on busy/no-answer, and it routes back here through SIP & RFC2833. |
| 00:22.48 | [TK]D-Fender | Ariel_ : SPA's allow me to choose in the GUI. works very well |
| 00:22.49 | Ariel_ | that is via sip not pri |
| 00:23.15 | [TK]D-Fender | Ariel_ : Just mentioning SIP as part of the equation, but not the part that impacts my server :) |
| 00:23.20 | [TK]D-Fender | errr PRI |
| 00:23.22 | Ariel_ | sipura yes lets you change it. But you if you set the sipura up as auto it works great as well. |
| 00:23.57 | [av]bani | De_Mon: dtmfmode defaults to rfc2833 if you dont define it afaict |
| 00:24.02 | [TK]D-Fender | Ariel_ : I don't trust anything that says "auto".... |
| 00:24.06 | Ariel_ | PRI is always via the Dchannel and since it's carried to the tdm network via the phone co it's inband. |
| 00:24.26 | [av]bani | yeah, it defaults to rfc2833 if not defined |
| 00:24.28 | [av]bani | just confirmed it |
| 00:25.15 | fdask | ive got an asterisk install running the demo dialplan. when i dial the first line of my card, asterisk picks up fine. but dialing the second line, asterisk doesn't even detect the call |
| 00:25.26 | fdask | i've tested the ports on the card using a util that comes with the driver, and that works fine |
| 00:25.46 | fdask | any reason why asterisk might not be recognizing the ring on the second port? |
| 00:27.18 | Mavvie | fdask: it shows up as a call with verbose set to say 5? |
| 00:27.36 | Ariel_ | set verbose 9 |
| 00:27.46 | Ariel_ | waht card? |
| 00:27.52 | Ariel_ | waht/what |
| 00:28.23 | fdask | openline4 |
| 00:28.26 | fdask | voicetronix |
| 00:28.35 | malverian[work] | Hmm.. is there a way to make Asterisk automatically work with nat=route and nat=yes hosts ? |
| 00:28.57 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 00:29.05 | Ariel_ | fdask, argh never set one up. |
| 00:29.41 | docelmo | Say anyone need a P4 Dell 1U or Supermicro Dual Xeon SuperServer? |
| 00:29.55 | fdask | ya, upping the verbose setting doesn't give any output when i call this second line |
| 00:29.57 | file[laptop] | yes, hand it over |
| 00:30.44 | fdask | when i start asterisk, the event log shows it initializing all 4 ports ok |
| 00:30.46 | fdask | :\ |
| 00:30.47 | *** join/#asterisk Grok_ (n=blah@c-71-196-75-163.hsd1.fl.comcast.net) |
| 00:30.47 | Ariel_ | docelmo, are you giving it away?? |
| 00:30.59 | tainted_ | what is openpbx? |
| 00:31.07 | Ariel_ | fdask, how did you set the channels up and context on the board |
| 00:31.27 | Ariel_ | tainted_, check out #openpbx it's a fork from asterisk |
| 00:31.38 | tainted_ | Ariel_ is it stable/any good? |
| 00:31.52 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 00:32.00 | Ariel_ | tainted_ this is not the right channel for openpbx |
| 00:32.12 | tainted_ | k |
| 00:32.26 | fdask | Ariel_: i set them up in vpb.conf |
| 00:32.42 | docelmo | Ariel_, sorry no.. Just liquidating the rest of my hosting company |
| 00:33.02 | Ariel_ | well then how much???? |
| 00:33.06 | tainted_ | docelmo what's the spec on the dell 1u |
| 00:33.12 | De_Mon | so, should I be using setCallerID or set(CALLERID(number)=name) ? |
| 00:33.23 | Ariel_ | which version |
| 00:33.25 | tainted_ | docelmo and how much are u asking |
| 00:33.27 | De_Mon | 1.2.2 |
| 00:33.40 | Ariel_ | De_Mon, 2nd one |
| 00:33.51 | docelmo | 600 Dell 2.8 1GB Ram and I would have to check the hard drives. either 80 or 200gb.. The supermicro is dual 2.8 xeon w/ 1gb and 120gb drive Im asking 1300 for it |
| 00:33.55 | Damin | docelmo: Fix your LCR engine yet? |
| 00:34.01 | De_Mon | Jan 24 19:29:11 ERROR[1483]: func_callerid.c:101 callerid_write: Unknown callerid data type. |
| 00:34.19 | tainted_ | docelmo which model.. the 850, 1850, 2850? |
| 00:34.20 | De_Mon | exten => _91NXXNXXXXXX,1,Set(CALLERID(8632973316)=Elephant|a) |
| 00:34.22 | docelmo | I created more headaches.. still working out the bugs of the new one.. but the problem we were experiencing are fixed |
| 00:34.25 | docelmo | 850 |
| 00:34.48 | tainted_ | ultraSCSI HDs? |
| 00:34.55 | alephcom_ | docelmo: if your db scheme is compatible, you're welcome to use my lcr engine. |
| 00:34.55 | tainted_ | or SATA |
| 00:34.56 | docelmo | No.. SATA |
| 00:35.10 | tainted_ | is it still under warranty? |
| 00:35.20 | docelmo | alephcom_, I can change it. . Where do I download it? Ya I think.. I would have to ckeck online |
| 00:35.38 | Damin | docelmo: You ready for me to stress test it for you? :) |
| 00:35.47 | docelmo | no.. not yet |
| 00:35.48 | tainted_ | alephcom_ what is your lcr engine api? webserice? |
| 00:36.28 | docelmo | tainted_ BJ Wickskey I cant spell his name.. but he bought 4 of em from me |
| 00:37.23 | tainted_ | how many do u have? are they dual proc? |
| 00:37.43 | docelmo | No P4 and 1 left. My supermicro is dual xeon |
| 00:37.48 | *** join/#asterisk pengyong (n=lala@222.188.133.19) |
| 00:38.20 | alephcom_ | http://www.astpp.org/index.php?n=Misc.LCR I have the lcr app written in C also and it runs as an asterisk app. DCC me if you want a copy. It'l be open source but I haven't got it all packaged yet. |
| 00:40.55 | *** join/#asterisk AlexCTI (i=AlexCTI@247.sub-70-197-31.myvzw.com) |
| 00:41.36 | *** join/#asterisk SwK (n=SwK@12-219-147-107.client.mchsi.com) |
| 00:43.22 | *** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net) |
| 00:44.22 | *** join/#asterisk posi211 (i=posi211@c-69-180-52-62.hsd1.ga.comcast.net) |
| 00:46.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:50.18 | malverian[work] | Strange.. |
| 00:50.43 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 00:53.17 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 00:53.57 | [TK]D-Fender | Why yes.. yes I am! |
| 00:54.51 | Ariel_ | [TK]D-Fender, hehe |
| 00:55.01 | fdask | heh |
| 00:55.03 | fdask | found my problem |
| 00:55.09 | fdask | channel numbering starts at 0 XD |
| 00:55.31 | fdask | the one line wasn't picking up because i didnt have a channel setup for it |
| 00:56.54 | [TK]D-Fender | Ariel_ : I use intermittant self-debasement to burn off karma :) |
| 00:57.40 | docelmo | Hay damin want a supermicro? |
| 00:57.51 | troyb1 | what are the specs ;) |
| 00:58.32 | docelmo | Dual 2.8 Xeon 1GB ram and I think 120gb HD |
| 00:58.32 | Ariel_ | [TK]D-Fender, since I am old I am no longer strange. I am excentric. |
| 00:58.34 | [TK]D-Fender | docelmo : What interesting is mine uses a i7205 chipset w/ E1000 NICs... 2 things that * HATES :D |
| 00:58.52 | docelmo | I ran asterisk on this box.. works fine.. |
| 00:58.59 | Ariel_ | docelmo, how much |
| 00:59.11 | [TK]D-Fender | Ariel_ : No its supposed to be "I'm not crazy..... poor people are crazy... I HAVE money.... I'm eccentric!" |
| 01:00.01 | docelmo | 1300 I paid 1800 a year ago for it. |
| 01:00.59 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 01:04.59 | *** part/#asterisk rend (n=rend@unaffiliated/rend/x-000000001) |
| 01:05.09 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 01:05.41 | Aughey | anyone familar with PoE? |
| 01:06.07 | Ariel_ | Power Over Ethernet ---- yes |
| 01:06.24 | NewSole | i am familar with P... I do it every morning I get up |
| 01:06.37 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 01:06.43 | tainted_ | Aughey what are u trying to do |
| 01:06.51 | tainted_ | PoE isn't overly complicated |
| 01:07.23 | Aughey | I have a client that would like to have their VoIP phone powered over the Ethernet. I know I need a PoE switch to provide the power on the lines, but I don't know what's required on the phone side |
| 01:07.24 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 01:08.03 | Ariel_ | Aughey, a phone that supports POE |
| 01:08.28 | Ariel_ | voipsupply has a new dlink 1526 for under 400 dollars I think that would make a great poe switch |
| 01:08.40 | tainted_ | Aughey most phones tell u if it supports PoE |
| 01:08.43 | Aughey | any suggestions for a phone for roughly $100? |
| 01:08.47 | tainted_ | lol |
| 01:09.01 | tainted_ | i think grandstream makes a phone around 100 |
| 01:09.02 | Ariel_ | poe hehehe under 100 bucks poe... |
| 01:09.26 | Aughey | so I take that as a no :) |
| 01:09.27 | tainted_ | actually poe phones are cheaper than their non-poe counterparts from what i've seen |
| 01:09.51 | [TK]D-Fender | Ariel_ : <400$ now? OMG |
| 01:10.03 | [TK]D-Fender | Price is CRASHING on these now... |
| 01:10.13 | tainted_ | what do u guys do for firewalls? |
| 01:10.17 | Ariel_ | [TK]D-Fender, yes and it's a pretty good working switch |
| 01:10.19 | tainted_ | sonicwall? |
| 01:10.19 | Aughey | they are a church, so cost is a big issue |
| 01:10.23 | Ariel_ | monowall |
| 01:10.24 | [TK]D-Fender | GRANDSUCK *shudder* |
| 01:10.38 | [TK]D-Fender | Ariel_ : I know, I bought 2 for $1000cdn ea |
| 01:10.47 | [TK]D-Fender | about 5 months ago |
| 01:11.11 | [TK]D-Fender | Aughey : what kind of setup are you looking to make? |
| 01:11.13 | Aughey | it would appear the GXP-2000 does support PoE and is roughly $100 |
| 01:11.32 | Ariel_ | Aughey, sorry but the two don't go hand and hand....Yes but remember you get what you pay for. |
| 01:11.35 | tainted_ | you might want to open it up and add some silly putty or clay |
| 01:11.49 | tainted_ | b/c it feels like a piece of crap in your hands |
| 01:11.50 | Aughey | approximately 30 phones. probably would need to power at least 10 |
| 01:11.57 | tainted_ | no thtat a church would care |
| 01:12.03 | [TK]D-Fender | Aughey : Granstream is really cheap stuff... don't think you want to run your company on those... |
| 01:12.22 | Zodiacal | if you don't have callerid from your phone co. is there still a delay of 2-3 rings in the usa? i have read that a delay answering and transfering etc.. can be due to this.. but i dont' have callerid.. i changed immediate in my zap configs to yes, and i still have the delay it seems.. :/ |
| 01:12.47 | Ariel_ | Zodiacal, yes unless you put in the zaptel to no |
| 01:13.05 | [TK]D-Fender | Zodiacal : That because you told it you had callerid on the line. the delay is automatic |
| 01:13.28 | Ariel_ | immediate should not be used unless your picking up a dumb system. |
| 01:13.29 | Zodiacal | i thought immediate=yes means no callerid support |
| 01:13.50 | Ariel_ | Zodiacal, no |
| 01:13.54 | [TK]D-Fender | Zodiacal : No thats for phones that ring a number immediately when you lift the receiver |
| 01:13.55 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 01:14.05 | Ariel_ | like door phones |
| 01:14.08 | [TK]D-Fender | Zodiacal : the "bat-phone" effect |
| 01:14.28 | Zodiacal | ew.. don't want that.. i was reading this post: http://forums.digium.com/viewtopic.php?t=2527&highlight=delay |
| 01:14.31 | Zodiacal | they said to do it |
| 01:14.32 | Zodiacal | :/ |
| 01:14.44 | Ariel_ | don't belive everything you read |
| 01:15.05 | Zodiacal | know of any options off hand that would cause my server to take a long time 5-10 seconds before dialing out? |
| 01:15.18 | Zodiacal | or even transfering incoming calls |
| 01:15.22 | Ariel_ | didgets nto matching?? |
| 01:15.30 | Ariel_ | nto/not |
| 01:15.40 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 01:16.13 | Zodiacal | i don't think i deal with did.. im new obvously, maybe i miss understood did |
| 01:16.33 | [TK]D-Fender | Zodiacal : pastebin yuor zapata.conf |
| 01:16.37 | [TK]D-Fender | ~pb |
| 01:16.38 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 01:16.39 | Ariel_ | Zodiacal, did are direct inward dials or numbers from the telco |
| 01:17.12 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 01:17.16 | Ariel_ | but if you look at things in order. like a menu. extensions 1xxx, then menu should start at 2,3,4,5 that way it does not have to match all the digets |
| 01:19.07 | Zodiacal | tkd-fender http://pastebin.ca/38470 |
| 01:19.37 | troyb1 | docelmo it sucks watching how fast gear deprecciates eh? |
| 01:19.52 | *** join/#asterisk lithi (n=interp3@toronto-HSE-ppp4045394.sympatico.ca) |
| 01:20.25 | Ariel_ | Zodiacal, ok your using amp. that looks ok but remove the immediate=yes to no... |
| 01:20.46 | [TK]D-Fender | Zodiacal : kill "usecallingpres=yes" and set immediate for no. I think you have a "fax" exten in your incoming context. that will do it to (STUPID AMP!!!) |
| 01:20.49 | Ariel_ | now how are your menu setup |
| 01:20.55 | [TK]D-Fender | FAX <----- |
| 01:20.57 | [TK]D-Fender | kill that |
| 01:21.44 | Zodiacal | in the zapa? |
| 01:21.50 | Zodiacal | or other configs? |
| 01:22.21 | Ariel_ | faxdetect=no |
| 01:22.24 | Ariel_ | is already there |
| 01:22.29 | Ariel_ | so no need to remove the fax |
| 01:23.07 | Zodiacal | should i rem useclalingpress or change it to no? |
| 01:23.47 | Ariel_ | usecallingpres=yes |
| 01:23.47 | Ariel_ | <PROTECTED> |
| 01:24.20 | *** join/#asterisk Leob (n=chatzill@wireless-19-37.media.mit.edu) |
| 01:24.20 | Aughey | can anyone suggest a vendor to purchase IP phones from? |
| 01:24.43 | Zodiacal | if i watch the FOP, it shows the ext and the trunk pick up, but the call counter on my phone doesn't start for like 5-10 seconds after that, then it starts to ring my test line... |
| 01:24.44 | Zodiacal | that make sence? |
| 01:25.31 | Ariel_ | Aughey, voipsupply.com |
| 01:25.56 | Ariel_ | how are your routes.... Zodiacal |
| 01:26.20 | Leob | Can anyone help me out? I'm getting : "chan_iax2.c:4967 authenticate_verify: requested inkey 'voicepulse01' for RSA authentication does not exist" What shall I do? |
| 01:26.42 | Ariel_ | install inkey from them |
| 01:26.50 | Leob | I did |
| 01:27.22 | Zodiacal | ariel even when i just use a simple ext. and dial() it takes a long time |
| 01:27.23 | *** join/#asterisk oceanlan (n=irc@cpe-69-133-109-130.woh.res.rr.com) |
| 01:27.26 | Leob | where does asterisk look for it? |
| 01:27.40 | Zodiacal | ariel how do you mean routes exactly? |
| 01:27.50 | Ariel_ | inbound routes |
| 01:27.59 | [TK]D-Fender | Ariel_ : I'm not sure that will stop the exten =fax,1, from kicking in.... |
| 01:27.59 | Ariel_ | inbound settings in the general section as well |
| 01:28.06 | *** join/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 01:28.14 | Ariel_ | [TK]D-Fender, it does not wait for that. |
| 01:28.28 | Ariel_ | there is only a wait(3) in the main rules |
| 01:28.37 | [TK]D-Fender | Aughey : www.atacomm.com beats voipsupply on the majority of * equipment... |
| 01:28.40 | zahid | hello everyone |
| 01:29.07 | Ariel_ | [TK]D-Fender, that fax is only used if it is detected. |
| 01:29.22 | zahid | noobie needs help registering asterisk as sip client with another sip-provider |
| 01:29.24 | Aughey | I've done most of my looking at voipsupply, but I'll check them both |
| 01:29.31 | [TK]D-Fender | Ariel_ : But to detect you have to wait the time between "bleeps" |
| 01:29.41 | Ariel_ | atacomm is good but I since I know the folks well at voipsupply I kinda like there service. |
| 01:29.48 | Zodiacal | ariel_ even when i dial out its slow.. i use amp for the inbound routes tho.. |
| 01:30.06 | Ariel_ | [TK]D-Fender, it's only wait(3) which is not even noticed |
| 01:30.14 | wunderkin | yeah i think atacomm has the cheapest prices on everything |
| 01:30.16 | Ariel_ | dial out via a sip phone |
| 01:30.23 | [TK]D-Fender | Ariel_ : I have been happy with voipsupply, but have heard lots of good about atacomm, and the price difference on a lot of important large items makes it worth it... |
| 01:30.45 | Ariel_ | if you call voipsupply on phone the match and beat atacomm |
| 01:30.51 | Ariel_ | but that is not for here |
| 01:31.00 | [TK]D-Fender | 30$ / savings on IP501 x 30 (his deployment) = $900 savings.... |
| 01:31.16 | [TK]D-Fender | Ariel_ : That would be a motivating factor... |
| 01:31.33 | Ariel_ | [TK]D-Fender, I get them for around 173... |
| 01:32.09 | Aughey | anyone have comments on the Grandstream GXP-2000? |
| 01:32.09 | zahid | service provider gives info for x-lite .. hostname for realm and sip proxy are different and not using port 5060 |
| 01:32.27 | zahid | having trouble getting asterisk to register |
| 01:32.33 | *** join/#asterisk [1]EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 01:32.55 | Ariel_ | zahid, ok what service and you can put what port you want on the sip.conf |
| 01:32.58 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 01:33.00 | Zodiacal | ariel any ideas on what could cause the delay on outgoing calls? |
| 01:33.12 | Ariel_ | Zodiacal, are you using a zip phone |
| 01:33.17 | [av]bani | http://bani.anime.net/phonez/ |
| 01:33.18 | Ariel_ | zip/sip |
| 01:33.24 | Zodiacal | ariel yeah cisco 7960g |
| 01:33.32 | Ariel_ | press the # key after your number |
| 01:33.45 | [TK]D-Fender | Ariel_ : $169 @ atacomm. |
| 01:33.48 | Ariel_ | that is the phones waiting for the matching |
| 01:34.13 | Zodiacal | ariel same.. theres a dial softkey that does that too |
| 01:35.14 | [TK]D-Fender | Aughey : Some people think they're the Holy Grail. A lot wish they never touched them (including a guy who's looking to rreplace 100 of them). |
| 01:35.14 | Ariel_ | then how is your outbound rules setup. |
| 01:35.14 | [TK]D-Fender | Most here think they are buggy and feel like crap. |
| 01:35.14 | zahid | ariel_ say if sip realm is sip-provider.com sip proxy is host.sip-provider.com:8551 |
| 01:35.14 | Aughey | that seems to be about par for the course |
| 01:35.15 | Zodiacal | ariel it dials to the server fast and picks up the trunk but it will ring like 5+ times before it really starts ringing on my incoming landline (real line to test) |
| 01:35.21 | Zodiacal | ariel_ i even tried dialing direclty just using dial() to the zap interface.. |
| 01:35.24 | Zodiacal | still delayed |
| 01:35.40 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 01:35.40 | Ariel_ | I would say it's your voip provider |
| 01:35.56 | *** join/#asterisk kiswanto (n=kiswanto@222.124.24.61) |
| 01:35.57 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 01:36.02 | Zodiacal | i don't have voip |
| 01:36.05 | Zodiacal | pots |
| 01:36.08 | Zodiacal | :P |
| 01:36.12 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 01:36.20 | Zodiacal | fxo's |
| 01:36.36 | kiswanto | help me please ! |
| 01:36.50 | kiswanto | how to convert wav into gsm file |
| 01:37.01 | [TK]D-Fender | kiswanto : "man sox" |
| 01:37.50 | kiswanto | I'm use sox with Open SuSE 10.0. but not support file type gsm. |
| 01:38.26 | Ariel_ | you don't have the complete sox install |
| 01:39.01 | Ariel_ | bbl baby needs me... (wondering why she is crying) |
| 01:39.18 | zahid | anyone knows which field in asterisk sip.conf is "domain/realm" of x-lite softphone? |
| 01:40.34 | wunderkin | zahid maybe fromdomain? i dont know a lot about sip, check the example config file |
| 01:40.42 | file | I'm running tail -f on the sineapps asterisk daily news site access log |
| 01:40.45 | file | it's very... amusing |
| 01:41.04 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 01:42.34 | *** join/#asterisk Shaquile (i=mattias@81.232.173.38) |
| 01:43.01 | Shaquile | What is the best (or a good) web based GUI for asterisk configuration? |
| 01:43.12 | zahid | wunderkin i tried that but no luck. |
| 01:44.45 | pifiu-laptop | hey fender and ariel |
| 01:44.50 | Err | xemacs |
| 01:44.51 | pifiu-laptop | i got it to work somewhat! |
| 01:45.00 | Err | oh, web-based - xemacs in a browser? |
| 01:45.04 | pifiu-laptop | i managed to get one way, just need to do the other way now |
| 01:45.40 | Ariel_ | pifiu-laptop, your extensions are the same on both sides. you should have one as a different number then the other |
| 01:46.19 | littleball | hello, does asterisk variable support float? |
| 01:46.39 | littleball | e.g., 1+1.0/20.0 |
| 01:48.11 | [TK]D-Fender | pifiu-laptop : Elaborate on "somewhat" |
| 01:49.08 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 01:49.23 | pifiu-laptop | give me one second guys, finishing up this email |
| 01:51.00 | *** join/#asterisk Jonathan_ (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 01:52.30 | Jonathan_ | Is anyone available to provide some troubleshooting assistance with Asterisk? |
| 01:52.46 | [TK]D-Fender | Jonathan_ : Always just ask your questions first and see what the reaction is |
| 01:54.45 | Jonathan_ | Ok, I am encountering an error when attempting to start asterisk. I am receiving: == Parsing '/etc/asterisk/codecs.conf': Found -- codec_lpc10: using generic PLC Ouch ... error while writing audio data: : Broken pipe Warning, flexibel rate not heavily tested! Segmentation fault and then it dies and returns to the bash prompt. |
| 01:55.19 | Jonathan_ | I'm not even sure where to start on this one and any help would be much appreciated. |
| 01:55.40 | littleball | hello, in the dial plan, will Hangup() trigger the "h" extension? |
| 01:56.03 | Aughey | is the echo cancellation worth it on the Digium Wildcard TDM2400? |
| 01:56.08 | [TK]D-Fender | littleball : No, it will simply hang up. "h" catches things before you choose to hang up |
| 01:56.26 | [TK]D-Fender | Aughey : VS none at all, sure. |
| 01:57.08 | Aughey | it'd be a $175 difference, so I'm checking all the options |
| 01:57.09 | littleball | [TK]D-Fender, pardon! which one is first? |
| 01:57.30 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 01:57.43 | littleball | [TK]D-Fender, i am a bit confused by the hangup . |
| 01:57.54 | [TK]D-Fender | littleball: I thought I was pretty clear. "h" gets triggered, Hangup gets CALLED. |
| 01:58.07 | SPoon_TSX | Hello everyone. I got a quick question on regarding the Echo Cancellation War between my TDM400 and PSTN. |
| 01:58.37 | [TK]D-Fender | littleball : "h" is used when someone hangs up when you are NOT expecting it. If you reach a part in your dialplan where you CHOOSE to hang up you should already have cleaned up after your call. |
| 01:58.58 | SPoon_TSX | I tried to get the ztmonitor to set my rxgain and txgain. When I start the testing, should it be the ##### level reach 50% or should be 100%? |
| 01:58.59 | Aughey | D-Fender: I assume you're serious about Sangoma |
| 01:59.12 | pifiu-laptop | ok so i did a complete overhaul |
| 01:59.23 | pifiu-laptop | i was pulling my hairs out from figuring all of this out |
| 01:59.33 | pifiu-laptop | let me tell you what i did |
| 01:59.54 | littleball | [TK]D-Fender, i am considering billing issue |
| 01:59.59 | [TK]D-Fender | Aughey : Rather. I've experienced a number of cards on both sides and my choice is clear. Compare the specs, read reviews and you be the judge. You know where to put my vote esp since EC was why I switched over to them. |
| 02:00.01 | *** join/#asterisk [2]EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 02:00.18 | SPoon_TSX | Any one? |
| 02:00.31 | [TK]D-Fender | littleball : Billing is a touchy matter which others can better advise on. I'v just given you the most of what I know of those 2 options. |
| 02:00.32 | littleball | [TK]D-Fender, "h" will also be triggered after hangup() called explicitly |
| 02:00.56 | [TK]D-Fender | littleball : Not to my memory, bt if you've read differently, then fine. |
| 02:01.13 | [TK]D-Fender | SPoon_TSX : Just go ahead and ask already! |
| 02:02.30 | SPoon_TSX | [TK]D-Fender: I just wondering when I use ztmonitor to set my rxgain and txgain. Should it be the Volumn level reach 50% of the bar or 100%? |
| 02:02.32 | littleball | [TK]D-Fender, Should i call Hangup() again if the "h" is triggered? |
| 02:03.32 | wunderkin | littleball no |
| 02:04.07 | pifiu-laptop | whats a good free softphone for testing basic stuff? |
| 02:04.30 | SPoon_TSX | pifiu-laptio: I would suggest X-lite |
| 02:04.33 | littleball | wunderkin, i try to ask the same question again. If hangup() is called explicitly, will it trigger "h"? |
| 02:04.40 | pifiu-laptop | yeah used xlite before |
| 02:04.48 | SwK | anyone around that works at a ITSP? |
| 02:04.48 | [TK]D-Fender | littleball : If you did that, wouldn't it just trigger "h" again? Circular logic I would think :) |
| 02:04.48 | pifiu-laptop | just wondering if anything else newer/better has come out |
| 02:05.11 | [TK]D-Fender | littleball : Hence "hangup" != "h" |
| 02:05.12 | littleball | [TK]D-Fender, yes. This is why i think i need to make it clear. |
| 02:05.19 | *** join/#asterisk tuxinator_linux (n=tuxinato@m110e36d0.tmodns.net) |
| 02:05.27 | SPoon_TSX | Another questions, is the rxgain=12 is a bit too high? |
| 02:05.32 | [TK]D-Fender | Hangup JUST HANGS UP. The end. Period. |
| 02:05.42 | [TK]D-Fender | SPoon_TSX : Thats absolutely psycho... |
| 02:05.52 | littleball | i also think so. That is to say, Hangup and 'h" have no relation |
| 02:06.07 | [TK]D-Fender | littleball : Like I said... |
| 02:06.07 | littleball | yes or no? |
| 02:06.12 | [TK]D-Fender | NONE. |
| 02:06.23 | SPoon_TSX | [TK]D-Fender: What will be the normal range? |
| 02:06.27 | Grubs | SPoon_TSX: my rxgain is=11. Verified with ztmonitor that its good (if not a bit low). |
| 02:06.31 | Aughey | [TK]D-Fender: Then with the Sangoma cards, is the Echo Cancellation option worth an extra $400 (for an 8 FXO Port) |
| 02:06.37 | littleball | [TK]D-Fender, then what is their relation? |
| 02:06.44 | [TK]D-Fender | "h" is why the caller disconnects whil you are processing things in the dial-plan. hangup is for when YOU kill him off when you're good and ready. |
| 02:07.05 | littleball | i need to prevent circular logic problem. and then make the billing correct |
| 02:07.24 | SPoon_TSX | Grubs: I got a quick questions, when i use the ztmonitor. Should I make my rxgain reach the 50% level or 100% when doing the milliwatt test? |
| 02:07.37 | wunderkin | littleball, im pretty sure that if you call hangup it will use the h exten but its easy enough to test, and no dont call hangup on h exten |
| 02:07.57 | [TK]D-Fender | Aughey : Absolutely. I have found Zaptel's SW EC quite inferior, and adds a load onto the server. I suffered horribly with my 2 TE405P, and once I switched to my A104d it instantly vanished. Not just a lot, I mean GONE |
| 02:08.05 | chrisb1 | 1/quit jjj |
| 02:08.13 | Grubs | Not sure about milliwatt. I aimed for 50% during fax send/receive and also voice calls |
| 02:08.43 | Grubs | verified the 50% on fax sending was right by sending to a test fax that reports the signal dB level. |
| 02:08.47 | littleball | [TK]D-Fender, i understand the basic concept . But i need info like wunderkin provided. The relation between them. |
| 02:08.54 | littleball | i will test it myself |
| 02:08.56 | [TK]D-Fender | SPoon_TSX : First of all you're playing around with gain's for EC? Set them to 0 to start, turn on basic EC (not echo training) and start in steps. |
| 02:09.07 | Aughey | [TK]D-Fender: Ok, I'm sold. Go-go gadget visa |
| 02:09.11 | [av]bani | Aughey: EC is the one thing you dont want to screw around with, hardware EC's are almost always superior |
| 02:09.19 | [TK]D-Fender | Then add echotraining in phases, then try another EC routine (recompile zaptel) |
| 02:09.33 | SPoon_TSX | Grubs: Do you mind to share your tunning experience on how to get rip the echo on the PSTN? Like how do you test it? |
| 02:10.08 | SwK | spoon_tx that really depends on what you are getting in |
| 02:10.12 | Aughey | I have the knockoff single FXO card and the echo is really bad |
| 02:10.12 | SwK | spoon_tx: i actually had to set tx_gain to -9 for a sprint dedicated LD channel once |
| 02:10.16 | SwK | seems their network is toon for 12db down end to end |
| 02:10.40 | Grubs | 1st - using X100P *nothing* removed echo so I junked the card and bought a TDM400P that could correclty match our australian line impedence. |
| 02:10.52 | SPoon_TSX | [TK]D-Fender: The problem I having now is in what standard should I consider that is good echo cancellation. Should I call a milliwatt number to test, or you have a better option? |
| 02:11.08 | [TK]D-Fender | Aughey : Things got better EC wise with 1.2, but I'd still prefer hardware EV every time. Whats great is there are a number of options in analog & PRI for it now. |
| 02:11.38 | SPoon_TSX | Grubs: May I know how you test your line with ztmonitor? |
| 02:11.42 | [TK]D-Fender | SPoon_TSX : The best test is a real call. The human ear is very picky and picks up things "tests" don't |
| 02:11.46 | *** join/#asterisk c4r3x (n=carex@202.159.57.221) |
| 02:12.04 | [TK]D-Fender | SPoon_TSX : Tuning EC can be a real pain. You are on 1.2.x now right? |
| 02:12.13 | SPoon_TSX | Swk: Woo.... So are you also using purely ztmonitor to setup your gain, no fxotune etc? |
| 02:12.19 | Grubs | sure. I simply run ztmonitor 1 -vv in a remote console while making a call or sending a fax |
| 02:12.40 | Aughey | [TK]D-Fender: One last question, then I'll warm up the Visa. I only need 6 ports right now. If I get the 6 port package, I could buy another daughterboard to bump it up to 8 ports easily, right. |
| 02:12.55 | SPoon_TSX | [TK]D-Fender: The real call, means I make a call and make sure every voice go thru my * box had 50% gain level? |
| 02:13.05 | [TK]D-Fender | Aughey : You're thinking of the A200 series? |
| 02:13.09 | Aughey | yes |
| 02:13.19 | Grubs | my rx goes about 1/3 up the scale, my tx goes about 2/3 up the scale. If I push rx any higher I get hiss on the line. |
| 02:13.20 | SPoon_TSX | [TK]D-Fender: I still on 1.0.6 |
| 02:13.27 | Aughey | or do you have another serious I should look at? |
| 02:13.29 | [TK]D-Fender | SPoon_TSX : no, the real call means you pick up a phone and CALL SOMEONE <- |
| 02:13.45 | [TK]D-Fender | SPoon_TSX : Upgrade to 1.2... zaptel EC has improved quite a bit <- |
| 02:14.01 | [TK]D-Fender | Aughey : No I just wasn't sure if you were looking for a PRI or not |
| 02:14.05 | Aughey | [TK]D-Fender: http://www.voipsupply.com/product_info.php?products_id=1341 |
| 02:14.08 | Grubs | I left zt monitor running - and then opoened a second remote console where I could tweak rxgain and txgain and reload - then make another test call. |
| 02:14.13 | litage | why does asterisk ship with such an old version of openh323? |
| 02:14.19 | SPoon_TSX | [TK]D-Fender: I cannot yet, since my server is not right beside me. |
| 02:14.45 | [TK]D-Fender | Aughey : Yes, if you get the A200 w/ HW EC with 6 ports, that 2 Remora w/ 3/4 filled. You have 1 module space free |
| 02:14.59 | [TK]D-Fender | SPoon_TSX : Doesn't need to be right beside you... but ok :) |
| 02:15.09 | [TK]D-Fender | SPoon_TSX : i've done all sorts of remote installs |
| 02:15.19 | Aughey | [TK]D-Fender: should should I purchase from? |
| 02:15.20 | Grubs | Spoon- see my posts here: http://forums.whirlpool.net.au/forum-replies.cfm?t=451077 |
| 02:15.28 | SPoon_TSX | Grubs: I got a funny things happen when I try to use ztmonitor 1 -vv command, it said "Unable to open /dev/dsp: No such device". |
| 02:15.30 | Grubs | (near the end of the thread) |
| 02:15.47 | [TK]D-Fender | Aughey : thats the one. 4 FXO / card = 1.5 cards. you can get another 2 ports really cheap. |
| 02:15.47 | Grubs | SPoon - is your ZAP channel "1"? |
| 02:15.54 | [TK]D-Fender | (later) |
| 02:16.00 | SPoon_TSX | [TK]D-Fender: So you suggest upgrade.. |
| 02:16.08 | SwK | spoon I use my ears |
| 02:16.22 | SPoon_TSX | Grubs: Acutally I got 4 FXO on my TDM400 card. |
| 02:16.25 | [TK]D-Fender | Aughey : Well the thing is atacomm and most other places haven't listed the A200 yet.... VoipSupply is the first to put it up. |
| 02:16.32 | [TK]D-Fender | (on this side of the ocean) |
| 02:16.37 | Jonathan_ | What does the codec_lpc10.so file provide to asterisk? |
| 02:16.38 | Aughey | that's what I noticed |
| 02:16.38 | [TK]D-Fender | SPoon_TSX : DEFINATELY. |
| 02:16.43 | [av]bani | oh heh, FXO... |
| 02:16.55 | [av]bani | 6 ports is almost breakeven for PRI in many places |
| 02:17.02 | [TK]D-Fender | SPoon_TSX : Startwith the base and work your way up. the primary EC routine has been seriously upgraded in 1.2 |
| 02:17.03 | [av]bani | and $800 would get you a 2 pri card |
| 02:17.21 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 02:17.21 | [TK]D-Fender | [av]bani : then again... what the cost of the SERVICE <- |
| 02:17.35 | [TK]D-Fender | [av]bani : And that card is HWEC. |
| 02:17.41 | pifiu-laptop | what is the correct port for SIP? |
| 02:17.43 | pifiu-laptop | so i can foward it |
| 02:17.47 | SPoon_TSX | [TK]D-Fender: Thank you for your suggest, but if I need to stay witht eh existing installation. What would be the steps you suggest me to take to eliminate the echo? |
| 02:18.05 | [TK]D-Fender | pifiu-laptop : 5060 for SIP, 10000-20000 for RTP. All UDP |
| 02:18.11 | SwK | damnit |
| 02:18.11 | SwK | my wife has a wifi jammer in the kitchen |
| 02:18.11 | SwK | she calls it a microwave |
| 02:18.15 | Grubs | Try just "-v" instead of "-vv" when a call is in progress perhaps. |
| 02:18.28 | *** join/#asterisk simprix (n=simprix@nowhere.simprix.net) |
| 02:18.43 | simprix | With my buttset how do i use groundstart on a demarc |
| 02:18.47 | SwK | jonathan_: it provides the codec LPC10 which makes you sound like mr roboto |
| 02:18.53 | [TK]D-Fender | SPoon_TSX : What I suggested earlier start from the base (0/0), and bas EC, then add echo training after in steps. then recompile to try other algorythms. |
| 02:18.55 | *** join/#asterisk Zachary[mIRC] (n=zachary@65.121.244.130) |
| 02:19.07 | SPoon_TSX | Grubs: If I use -v, then it works and give me the bar. I also patched that show me the actual number of the volumne too. |
| 02:19.08 | Zachary[mIRC] | can anyone help me with AMP |
| 02:19.11 | *** join/#asterisk VirtualUK (n=newkinet@shell2.sea5.speakeasy.net) |
| 02:19.12 | SwK | simprix: use a piece of wire connected to ground to start the loop |
| 02:19.18 | Grubs | SPoon_TSX: is your TDM card running in the right "opermode" for your country? |
| 02:19.29 | Jonathan_ | Swk: So I can disable it without a problem then? Asterisk crashes whenever it tries to load the module. |
| 02:19.31 | [TK]D-Fender | Zachary[mIRC] : try #amportal |
| 02:19.37 | [av]bani | [TK]D-Fender: interfacing to analogue PSTN sux, if it can be avoided its highly recommended |
| 02:19.37 | SwK | spimprix: ie: use a piece of jumper wide connected to a ground then touch it to one side |
| 02:19.43 | SPoon_TSX | [TK]D-Fender: base EC means make every echoxxxx=no right? |
| 02:19.48 | litage | does asterisk use threads to take advantage of multi-cpu servers? |
| 02:19.53 | SwK | jonathan yes no problems there |
| 02:20.03 | [TK]D-Fender | [av]bani : I'd agree, but try and compare the cost when a person only needs 6 lines <- |
| 02:20.08 | simprix | Swk: there is a groundstart button on the phone how can i use it |
| 02:20.16 | Jonathan_ | SwK: Thanks much. Talk about an easy solution. |
| 02:20.16 | SPoon_TSX | Grubs: I am in North America, Canada. I hope is still consider US Standard? |
| 02:20.18 | Zachary[mIRC] | [TK]D-Fender ther eis no one there to help :s |
| 02:20.23 | [av]bani | [TK]D-Fender: well, in many places 6 lines is breakeven, and you get a lot more functionality with PRI |
| 02:20.28 | Grubs | lol - should be ok then |
| 02:20.45 | [TK]D-Fender | SPoon_TSX : No, it means "echocancel=yes", "echotraining=no", "rxgain=0.0", "txgain=0.0" |
| 02:21.02 | SwK | simprix: sure |
| 02:21.08 | SPoon_TSX | Grubs: So, when you trim your rxgain and txgain. Which one you train first? rxgain or txgain? |
| 02:21.11 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 02:21.14 | [TK]D-Fender | SPoon_TSX : try that first, then add in "echotraining=yes", then in #'s in intervals. All before touching your gains. |
| 02:21.25 | simprix | Swk how |
| 02:21.42 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:21.50 | [TK]D-Fender | [av]bani : 6 analog VS a PRI? Never seen PRI Anywhere near that closely priced. |
| 02:22.10 | VirtualUK | I know this question comes up a ton, but I'm stuck installing my X100P clone (Tiger Jet 3xx) with the usual problem "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" when executing the "modprobe wcfxo" command.....I've tried switching PCI slots and most of the other suggestions that I found by searching......anyone else have any ideas? |
| 02:22.11 | SwK | simprex: well i assum your buttset has to be grounded somehow then you take it off hook and push the groundstart button |
| 02:22.17 | SPoon_TSX | [TK]D-Fender: OIC, in other words I should try all of the echoxxx command, before I touch the gain settings. May i know which EC Method you used the most ini your zconfig.h? |
| 02:22.22 | [av]bani | [TK]D-Fender: new jersey, hell atlantic will do PRI for like $150 |
| 02:22.26 | Grubs | I just have the default echotraining=yes. I tweaked the gains to get the levels about right. Echo was never a problem. |
| 02:22.27 | SwK | simprex: else just use the grounded wire method that what most people use |
| 02:22.40 | [TK]D-Fender | VirtualUK : Do you see it in cat /proc/interrupts? |
| 02:22.48 | littleball | wunderkin, you are right. Hangup() also trigger "h". |
| 02:22.56 | VirtualUK | TK: Nope, but I do see it when I do an lspci |
| 02:23.31 | SPoon_TSX | Grubs: oic, let me give my * another shot to see if it works out. Wish me good luck. |
| 02:23.32 | [TK]D-Fender | SPoon_TSX : exactly. playing with gains is the BS way of trying to cheat things because you didn't really fix them. so you try to lessen what returns by lessening what you send... impacts the other side a lot too... |
| 02:24.00 | [TK]D-Fender | [av]bani : $150 for a PRI? How many channels? |
| 02:24.35 | [av]bani | [TK]D-Fender: 23, and then like $5 per did |
| 02:24.47 | justinu | nice |
| 02:24.53 | SPoon_TSX | [TK]D-Fender: Thanks for your help. Let me try it out now, and wish my good luck. |
| 02:24.55 | [av]bani | tis why so many boiler rooms operate out of NJ |
| 02:25.05 | [TK]D-Fender | $150 for a 23 port full PRI? I seriously can't believe that..... |
| 02:25.30 | [TK]D-Fender | Aughey : Have you researched PRI rpicing in your area? |
| 02:25.46 | *** join/#asterisk minsu (n=minsu@59.12.193.2) |
| 02:25.55 | [av]bani | [TK]D-Fender: neither could i, but apparently thats what my friend's company pays |
| 02:26.02 | [av]bani | maybe its not BA but a clec |
| 02:26.12 | [TK]D-Fender | [av]bani : HMMMM :/ |
| 02:26.28 | Aughey | that's not really an option for us. We have 4 lines from a university, and 2 lines from Southwestern Bell. If the univeristy would provide it, we probably would |
| 02:26.43 | VirtualUK | [TK]D-Fender : it doesn't show up in /proc/interrupts but it does show up with an lspci as "00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" |
| 02:26.45 | [TK]D-Fender | Aughey : Ok, you're set then! |
| 02:27.02 | [av]bani | if you dont need 5 9's you could get voip trunks :D |
| 02:27.12 | Aughey | where would I find out about PRI? |
| 02:27.17 | [TK]D-Fender | VirtualUK : Well..... thats a tough one. |
| 02:27.29 | [TK]D-Fender | Aughey : ask local telco's what it'd cost. |
| 02:27.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 02:28.22 | VirtualUK | [TK]D-Fender : I'm assuming that I should see some wcfxo entry in /proc/interrupts....is that true even before zaptel and wcfxo are loaded? |
| 02:28.35 | justinu | no |
| 02:29.01 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 02:29.28 | VirtualUK | was that to my question justinu? |
| 02:29.40 | justinu | yeah |
| 02:29.46 | [TK]D-Fender | Aughey : And yes, there is the option of straight VoIP trunks. have you considered this? |
| 02:29.50 | VirtualUK | :) thanks |
| 02:30.35 | VirtualUK | justinu : so the wcfxo in /proc/interrupts will only appear after a successful execution of modprobe wcfxo right? |
| 02:30.51 | justinu | i believe so |
| 02:30.57 | Aughey | [TK]D-Fender: you mean VoIP on the Internet, right? |
| 02:31.26 | [TK]D-Fender | Aughey : Correct. Using phone numbers provided to you over the internet. |
| 02:31.52 | Aughey | we might consider that for optional connections, but I'm not convinced of their reliability. |
| 02:32.13 | [TK]D-Fender | Aughey : If its a concern for you then go right ahead with analog. |
| 02:32.45 | Aughey | I might call to see about the PRI, the University might perfer to supply that instead of 4 separate analog lines |
| 02:33.16 | [TK]D-Fender | Aughey : Depending on your arrangement you can do a lot more with it and perhaps enhance their lives as well. |
| 02:34.10 | *** join/#asterisk carol (n=claudia_@200.58.81.58) |
| 02:34.22 | [TK]D-Fender | Aughey : How much is a line w/ callerID where you are? |
| 02:34.26 | VirtualUK | anyone else in here using a cloned X100P? |
| 02:35.03 | Aughey | It's going to be SBC.... |
| 02:36.06 | *** join/#asterisk Chogun (n=fabienec@53530A13.cable.casema.nl) |
| 02:36.20 | Chogun | je cherche obelix |
| 02:36.36 | Chogun | quelqun l'aurais vu par hasard ? |
| 02:36.43 | Aughey | It's complicated, becase we get DSL through SBC and use it for Fax and unlimited long distance dialing. Then we have the 4 lines from the university for our main dial-in numbers |
| 02:36.44 | *** part/#asterisk Zachary[mIRC] (n=zachary@65.121.244.130) |
| 02:36.58 | Aughey | VirtualUK: I have one |
| 02:37.07 | Chogun | ça me dis pas ou ce trouve obelix... |
| 02:37.36 | [TK]D-Fender | Chogun : Chercez ailleurs 'st... |
| 02:37.44 | [TK]D-Fender | sti* |
| 02:37.46 | [TK]D-Fender | :D |
| 02:38.01 | [TK]D-Fender | ... c'est a dire.. va t'ens! |
| 02:38.21 | VirtualUK | aughey : any chance you could email me your zaptel, zapata and asterisk conf files? this damn error is driving me crazy! :) |
| 02:38.43 | [TK]D-Fender | Aughey : You have 4 analog lines, not 6? |
| 02:39.02 | Aughey | don't have them available right now. That machine is being formatted as we speak and I'm jumping ship with the X100P's :) |
| 02:39.13 | Aughey | [TK]D-Fender: 4 from the university and 2 from SBC |
| 02:39.20 | [TK]D-Fender | Aughey : AHH |
| 02:39.34 | [TK]D-Fender | Aughey : not sure how much they cost you? |
| 02:39.42 | Aughey | I don't handle the billing |
| 02:40.23 | troyb1 | Aughey its better that way :) |
| 02:40.28 | Aughey | the only "pure" line is the fax line. The other SBC has DSL and unlimited long distance |
| 02:40.59 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 02:41.02 | *** join/#asterisk muzzz (n=chatzill@218.111.151.65) |
| 02:42.31 | [TK]D-Fender | Aughey : You don't know what it costs you? |
| 02:42.47 | Aughey | no |
| 02:42.57 | [TK]D-Fender | :/ |
| 02:43.29 | [TK]D-Fender | Aughey : I like being fully aware of costs, current and projected for all my options before spending the kind of cash you are. |
| 02:43.54 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 02:44.17 | [TK]D-Fender | Aughey : Thogh I doubt you'll have any regrets for the nature of what you're looking at, its the unknown bit that irks me. |
| 02:46.53 | Aughey | I also need some FXO ports to tie into an existing phone system. That's only for a short time until the renovation is complete |
| 02:47.24 | *** join/#asterisk SDGL (n=sdgl@64.5.206.131) |
| 02:47.27 | SPoon_TSX | [TK]D-Fender: Hi there, I almost get rip all of the echo. May i know what is the normal setting for echotraining? Can I set it to 60? Would it be too low? |
| 02:48.19 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 02:49.49 | [TK]D-Fender | SPoon_TSX : Not sure on the fine points of those settings. Take a look at the WIKI on that one. and start in small steps |
| 02:49.51 | lithi | Any idea how I could make a callback feature such that a user dials a number and will get a busy signal then (using calleridnum) asterisk waits 5 seconds then calls the person back? Im sure someone already has a AGI script or something for this.. just cant find it using google |
| 02:50.27 | [TK]D-Fender | Aughey : ok, Sounds like thats the tech for your scenario. Its always resellable later whne you change your mind I guess... |
| 02:50.43 | SPoon_TSX | [TK]D-Fender: What about the echocancel, which one you use the most 64, 128 etc? |
| 02:51.35 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 02:51.56 | *** join/#asterisk bch (n=bch@CPE-70-92-133-175.mn.res.rr.com) |
| 02:52.08 | smallb | anybody using the digium g729 codec ? |
| 02:52.24 | *** part/#asterisk carol (n=claudia_@200.58.81.58) |
| 02:53.12 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 02:54.01 | pifiu-laptop | enough * for today for me |
| 02:54.02 | pifiu-laptop | seriously |
| 02:54.05 | pifiu-laptop | i cant take it |
| 02:54.23 | Aughey | [TK]D-Fender: and $800 isn't too bad to spend. If I need more, I can add more cards. If the incoming lines change in the future, no big deal. |
| 02:56.08 | [TK]D-Fender | SPoon_TSX : right now I don't use any. My SPA-3000 has its own (G.168-2000 which is so/so), and my work is a Sangoma A104d (G.168-2002 which is godly). |
| 02:56.27 | [TK]D-Fender | SPoon_TSX : I only used basic echotraining on my old TDM22b before ditching it. |
| 02:56.36 | [TK]D-Fender | Aughey : Yup. |
| 02:56.51 | SPoon_TSX | [TK]D-Fender: You mean you set echocancel=no? |
| 02:57.15 | [TK]D-Fender | SPoon_TSX : no, I mean I did echocancel=yes and echotraining=yes at most at home |
| 02:57.37 | [TK]D-Fender | I tried everything imaginable to fix the echo on my TE405P to no avail so I ditched it. |
| 02:57.53 | [TK]D-Fender | pifiu-laptop : So got it working? |
| 02:57.57 | *** join/#asterisk unni (n=chatzill@web01.nccmortgage.co.jp) |
| 02:58.13 | Aughey | [TK]D-Fender: you've been a big help tonight. Thanks. |
| 02:58.56 | pifiu-laptop | fender, one way |
| 02:58.57 | [TK]D-Fender | Aughey : Glad to help. |
| 02:59.03 | pifiu-laptop | go figure |
| 02:59.11 | [TK]D-Fender | pifiu-laptop : One-way means you're half way there! |
| 02:59.14 | pifiu-laptop | you would think yay perfect, just do the same on the other machine |
| 02:59.28 | pifiu-laptop | but it just seemed to easy and didnt work |
| 02:59.33 | pifiu-laptop | are you online tomorrow? |
| 02:59.42 | [TK]D-Fender | pifiu-laptop : both servers accessable to th net? |
| 02:59.47 | pifiu-laptop | yes they are |
| 03:00.08 | [TK]D-Fender | PM me the details and I'l see about fixing in here and now. Should be quick |
| 03:05.30 | *** join/#asterisk jef_ (i=fischer@p54846E6E.dip.t-dialin.net) |
| 03:05.41 | *** join/#asterisk rend (n=rend@unaffiliated/rend/x-000000001) |
| 03:06.31 | SPoon_TSX | Grubs: If I can hear myself, should I adjust the rxgain or txgain? |
| 03:06.37 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 03:07.41 | [av]bani | txgain |
| 03:07.45 | [av]bani | lower it |
| 03:09.21 | SPoon_TSX | [av]bani: lower the txgain right? |
| 03:09.36 | *** part/#asterisk VirtualUK (n=newkinet@shell2.sea5.speakeasy.net) |
| 03:09.37 | Aughey | [TK]D-Fender: ok, ordered the 6 port FXO model. And a Grandstream GXP-2000. |
| 03:10.49 | [av]bani | yes |
| 03:11.35 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-156-177-71.mia.bellsouth.net) |
| 03:11.44 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-156-177-71.mia.bellsouth.net) |
| 03:12.41 | SPoon_TSX | [av]bain: May I ask can the echo be completely eliminated? And for txgain, how low will be too low? |
| 03:13.17 | Math[laptop] | SPoon_TSX: do you have ztmonitor? |
| 03:13.33 | Math[laptop] | er, you have it, run it on a zap channel and make a few calls |
| 03:13.40 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 03:13.47 | Math[laptop] | then adjust the rx & tx levels to get as close as possible to the middle |
| 03:14.33 | SPoon_TSX | Math[laptop]: You mean every single words I say should make the graph close to the middle? |
| 03:14.55 | Math[laptop] | SPoon_TSX: well its a volume metter, it shouldnt be too high or too low |
| 03:15.34 | SPoon_TSX | Math[laptop]: How you test it? Like call someone and chat with them or you have a special number to call? |
| 03:17.34 | *** join/#asterisk remmos (n=chatzill@adsl-1180.isp.net.au) |
| 03:18.06 | *** join/#asterisk VirtualUK (n=newkinet@shell2.sea5.speakeasy.net) |
| 03:18.22 | *** join/#asterisk FastJack_ (i=fastjack@p5091FD27.dip.t-dialin.net) |
| 03:18.40 | Math[laptop] | SPoon_TSX: call & chat :) |
| 03:19.00 | VirtualUK | alrigty then.....I think I'm a lil closer to figuring out my cloned x100p problem. When I do a lspci -vv it shows the card on IRQ 47 ..... that doesn't look right, can anyone confirm? |
| 03:19.33 | VirtualUK | I didn't think IRQ's went that high? |
| 03:20.11 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 03:20.47 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:21.07 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 03:22.14 | Err | VirtualUK: IRQs on the PCI bus don't go that high, no, but modern interrupt controllers can map interrupts all over the place |
| 03:25.11 | *** join/#asterisk iq (n=iq@71-214-1-78.omah.qwest.net) |
| 03:25.22 | VirtualUK | can the irq of a card be reassigned? |
| 03:28.48 | Err | not exactly - a PCI IRQ is routed to the interrupt controller, which then raises an interrupt with the CPU - and then the interrupt controller can choose to identify that interrupt as whatever value it wants, within whatever range it supports |
| 03:29.05 | Err | IRQ numbers aren't really relevant these days |
| 03:29.24 | Err | the only concern is when you have two devices with the same PCI or logical IRQ, really |
| 03:30.02 | VirtualUK | Err: what's odd is that my x100p clone (tiger jet 3xx) card shows up under lspci, but not when I cat /proc/interrupts |
| 03:32.59 | Err | is it not working? |
| 03:33.02 | rend | why is every other page in the asterisk book fucked up? |
| 03:33.13 | rend | i mean every other chapter in the pdf |
| 03:33.21 | rob0 | Some motherboards do allow setting PCI IRQ's, no? |
| 03:33.24 | Err | if you're referring to the AsteriskTFOT.pdf file, it isn't for me |
| 03:33.29 | zahid | hello all. anyone familiar with sip client auth using digest/md5 challenge? |
| 03:33.43 | zahid | using realm |
| 03:34.09 | VirtualUK | Err : nope, when I do a modprobe wfxo I get the ZT_CHANCONFIG error |
| 03:35.06 | rend | Err: yes. thats what i am talking about.. using gnome pdf viewer |
| 03:35.20 | Err | rob0: yes - but that's Yet Another Topic :-) |
| 03:35.29 | rob0 | YAT! |
| 03:35.32 | Err | rend: works fine with me in xpdf, on both OSX and Linux |
| 03:35.46 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 03:36.01 | *** join/#asterisk muzzz_ (n=chatzill@218.111.65.17) |
| 03:38.45 | rend | Err: isnt as good as gnome's but at least it renders all the pages properly, thanks |
| 03:39.46 | *** join/#asterisk muzzz_ (n=chatzill@218.111.65.17) |
| 03:41.50 | dmz | anyone know a good url/reference on asterisk trunking? i want to trunk my home box to my colo box |
| 03:42.20 | rend | has anyone here used asterisk with axvoice? i want to get it to answer my calls... |
| 03:42.30 | Err | it's 100,000x better than gnome's |
| 03:42.57 | rend | Err: good |
| 03:43.29 | *** join/#asterisk copantl (n=copantl@205.240.200.82) |
| 03:43.29 | Math[laptop] | argh I'm getting loopback dundi information |
| 03:45.00 | ke4qqq | Has anyone ever seen asterisk interpret a dtmf string with an |
| 03:45.09 | ke4qqq | 'i' in it? ala: http://pastebin.ca/38479 |
| 03:48.03 | znoG | VirtualUK: have you edited /etc/zapata.conf? |
| 03:48.13 | znoG | VirtualUK: i mean /etc/zaptel.conf |
| 03:48.30 | SPoon_TSX | Rrrrrrr.... I almost give up. No matter, How I tune, I can still bearly here myself talking.... My rx=0.0 tx=-2.0, echocancel=64. Any idea? |
| 03:49.25 | SPoon_TSX | Any experts have any idea?? Please help! |
| 03:49.54 | *** join/#asterisk bmg505 (n=leon@dsl-146-35-67.telkomadsl.co.za) |
| 03:52.57 | *** join/#asterisk budmang (i=budman@12-216-205-236.client.mchsi.com) |
| 03:53.02 | budmang | :-) |
| 03:53.11 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 03:53.18 | Dr-Linux | question, i have 2 asterisk servers, i wanna get call on 2nd server |
| 03:53.23 | Dr-Linux | exten => _3XXXXXXXXXX,1,Dial(SIP/202.125.141.11/${EXTEN:1},30,r) |
| 03:53.27 | budmang | will someone call me through my asterisk? |
| 03:53.49 | *** part/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 03:53.50 | Dr-Linux | i did this, so i wanna recieve call on 202.125.141.11 server |
| 03:53.52 | Dr-Linux | any idea? |
| 03:55.16 | Dr-Linux | currently its getting error |
| 03:55.16 | Dr-Linux | -- SIP/202.125.141.11-4eef is circuit-busy |
| 03:55.16 | Dr-Linux | <PROTECTED> |
| 03:55.17 | Dr-Linux | <PROTECTED> |
| 03:55.23 | NewSole | Dr-Linux... that dial scrip u are passing passes no username or pass |
| 03:55.32 | SPoon_TSX | Math[laptop]: May i know what sure the graph show on the rx when no one making a call? |
| 03:55.37 | SwK | Dr-linux: set them up as friends in sip.conf [servera] type=friends host=ip insecure=very context=your_dialplan_context sip reload |
| 03:55.42 | dmz | dr-linux use iax between the 2 |
| 03:55.44 | SPoon_TSX | Math[laptop]: Should it be say 0? |
| 03:55.45 | NewSole | you will get "Everyone is busy/congested at this time" cause it can not auth |
| 03:55.53 | dmz | i'm trying to get trunking setup to do that same thing |
| 03:55.59 | SwK | dr-linux: the dial(sip/servera/${EXTEN}) or whatver |
| 03:56.01 | dmz | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 03:56.05 | budmang | Can anyone help me with NAT? |
| 03:56.15 | budmang | Ive got 4 people with ext. that work fine |
| 03:56.20 | budmang | but sometimes i cant conenct |
| 03:56.23 | budmang | or people cant call me |
| 03:56.25 | NewSole | I heard RAID is good with NAT's |
| 03:56.32 | *** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz) |
| 03:56.35 | Psykick | hi guys |
| 03:56.35 | Dr-Linux | :S |
| 03:56.42 | Dr-Linux | SwK: can i /msg you ? |
| 03:56.58 | *** join/#asterisk doofoo (n=irc@68.238.167.98) |
| 03:57.08 | Psykick | I hope these new QoS switches we have improve the quality of the calls we're making tonight |
| 03:57.27 | SwK | dr-linux: why not ask in public looks lime DMZ is having similar issues |
| 03:57.47 | Dr-Linux | but my english |
| 03:57.48 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 03:57.51 | Dr-Linux | okey let me t ry SwK |
| 03:57.55 | budmang | Im in the DMZ but still having problems would NAT fix this? |
| 03:57.59 | Psykick | has anyone here using asterisk in a large scale operation? like a call centre? |
| 03:58.03 | Dr-Linux | SwK: on serverA >> set them up as friends in sip.conf [servera] type=friends host=ip insecure=very context=your_dialplan_context sip reload |
| 03:58.04 | Dr-Linux | in sip |
| 03:58.09 | Dr-Linux | and what about ServerB ? |
| 03:58.17 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 03:58.21 | SwK | same thing on server b |
| 03:58.28 | SwK | sorry bout that |
| 03:58.41 | NewSole | Psykick... whats the problem |
| 03:58.53 | SwK | psykick yes |
| 03:58.53 | Psykick | get a variety of issues with call quality |
| 03:58.59 | SwK | several of us do |
| 03:59.01 | NewSole | I am here for a few min... could not sleep having a smoke break |
| 03:59.01 | Psykick | we're using uLaw at the moment |
| 03:59.04 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 03:59.09 | NewSole | ACK |
| 03:59.13 | SwK | psykick are you overloading the network |
| 03:59.20 | Psykick | quite possibly |
| 03:59.21 | NewSole | i would say so |
| 03:59.30 | Psykick | that's the reason we got the QoS switches in |
| 03:59.37 | budmang | MeetMe doesnt work either? Can anyone help me? or point me in the right direction |
| 03:59.46 | Psykick | they've been configured to give highest priority to UDP 4569 |
| 03:59.53 | NewSole | QOS is going to do nuthin.... |
| 03:59.53 | SwK | psykick: theres the problem |
| 04:00.10 | Psykick | ok so what should I do about it? |
| 04:00.14 | NewSole | whats your network speed Psykick |
| 04:00.17 | SwK | heavily loaded IAX almost always has audio qualit issues |
| 04:00.22 | Psykick | at the moment it's 100mbps |
| 04:00.23 | SwK | switch to sip and be done with it |
| 04:00.41 | Psykick | we can go 1000mbps but need to get different switches |
| 04:00.57 | NewSole | whats your server config |
| 04:01.03 | Psykick | pretty basic |
| 04:01.09 | znoG | how is switching to SIP going to help though? sounds like the media path traffic would be the only thing causing a problem |
| 04:01.14 | Psykick | everyone has an IAX2 extension |
| 04:01.15 | NewSole | and how many calls at once |
| 04:01.22 | unni | <PROTECTED> |
| 04:01.26 | Psykick | at most probably 20 - 25 |
| 04:02.13 | Dr-Linux | SwK: just wanted to confirm |
| 04:02.17 | NewSole | well do the math problem = (64k + 22k) * 20 |
| 04:02.29 | Dr-Linux | SwK: is that ok for serverA ? |
| 04:02.34 | SwK | yeah like that |
| 04:02.45 | SwK | you can allow add permit /deny lines |
| 04:02.46 | file | er |
| 04:02.48 | file | SwK! |
| 04:02.50 | SwK | to filter on IP |
| 04:02.53 | SwK | file! |
| 04:02.55 | Dr-Linux | okey let me do on serverB |
| 04:02.58 | Dr-Linux | like? |
| 04:03.00 | pifiu-laptop | so people using * in call centers use what? |
| 04:03.37 | NewSole | whats your feeds.... softphone or pri's |
| 04:03.44 | Psykick | NewSole: what's the 22k? the audio sample rate? |
| 04:04.03 | NewSole | tcp overhead |
| 04:04.11 | SPoon_TSX | Can anyone share their Echo hunting experience? I cannot get rip of my echo problem. No matter which number I tried, I can still bearly hear myself. Is that normal? |
| 04:04.16 | Dr-Linux | SwK: question, on the serverA i'll put serverB ip address right in host=ip field ? |
| 04:04.22 | SwK | yep |
| 04:05.19 | znoG | how is switching to SIP going to help though? sounds like the media path traffic would be the only thing causing a problem |
| 04:05.31 | *** join/#asterisk sprout (n=sproutin@67-41-102-122.albq.qwest.net) |
| 04:05.37 | *** part/#asterisk sprout (n=sproutin@67-41-102-122.albq.qwest.net) |
| 04:05.42 | znoG | SPoon_TSX: played with the rxgain/txgain settings i take it? |
| 04:06.10 | znoG | SPoon_TSX: its important to understand where/how echo is produced. There is some info on this on the voip-info site which also includes a link to an interesting document written by Cisco (from memory) about echo cancellation. |
| 04:06.44 | Dr-Linux | SwK: thats serverB setting |
| 04:06.46 | *** join/#asterisk Drew (n=foo@zux221-065-169.adsl.green.ch) |
| 04:06.53 | Dr-Linux | now what about dialplan ? |
| 04:06.59 | SPoon_TSX | I spent 4 hours to play with it, no matter how I try. I still have a very minder echo issue. I am just wondering is that echo cannot be 100% eliminated on PSTN? |
| 04:07.12 | Dr-Linux | exten => _3XXXXXXXXXX,1,Dial(SIP/202.125.141.11/${EXTEN:1},30,r) |
| 04:07.15 | Dr-Linux | is it okey? |
| 04:08.09 | Drew | is that ip your sip provider? @drlinu |
| 04:08.19 | znoG | SPoon_TSX: what sort of FXO card? |
| 04:08.37 | SPoon_TSX | TDM400, with 4 FXO. Am I just bad luck? |
| 04:09.12 | znoG | i have the same. Have you tried rxgain=4 and txgain=8 ? |
| 04:09.15 | znoG | i think thats how I had it setup |
| 04:09.20 | znoG | or txgain=8 rxgain=4, can't remember |
| 04:10.01 | SPoon_TSX | Where are you located? |
| 04:10.06 | SwK | are you sure the echo is coming from the TDM channels in the first place? if you are, are you sure you need higher gains and not lower gains |
| 04:10.15 | znoG | SPoon_TSX: in Argentina |
| 04:10.51 | SPoon_TSX | Swk: I am sure. Since SIP call have no problem and regular phone has no problem only the call via * over the TDM has the problem. |
| 04:10.57 | SPoon_TSX | znoG: Let me try. |
| 04:11.29 | SwK | spoon_tsx: just checking echo's a bitch to troubleshoot sometimes |
| 04:11.32 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 04:12.05 | SPoon_TSX | Swk: I think I may need to give up. |
| 04:12.20 | SPoon_TSX | Swk: What is your settings? May i take it as a reference? |
| 04:12.30 | SwK | i dont have one here |
| 04:12.41 | SPoon_TSX | Swk: From you memory? |
| 04:12.45 | SwK | and the rxgain/txgain settings vary from deployment to deployment |
| 04:13.33 | SPoon_TSX | Swk: Is that true that if I increase the rx I should also decrease the tx? |
| 04:13.43 | SwK | no |
| 04:13.59 | SwK | they are independant settings |
| 04:14.12 | SPoon_TSX | So you mean, they could be tx > rx? |
| 04:14.16 | SwK | you should adjust them up and down independantly as required |
| 04:14.18 | SwK | yea |
| 04:14.24 | Defraz | Hey I got my little astrisk box provisioned to Teliax, works great but I can't seem to get caller ID working |
| 04:14.28 | Defraz | I have a pap2 |
| 04:14.44 | Defraz | Any place to look and see why? |
| 04:14.48 | SPoon_TSX | Swk, how do I tell when do I need to adjust the tx and when to rx? |
| 04:14.55 | SwK | i've cussed bell over shitty gain settings on their end in a previous life dealing with 4wire dedicated circuits |
| 04:15.03 | SwK | call someone and listen to it |
| 04:15.03 | *** part/#asterisk VirtualUK (n=newkinet@shell2.sea5.speakeasy.net) |
| 04:15.08 | Drew | i have a problem with creating menus in the extensions.comf: - i made a menu so incoming calls could choose which phone they want to ring - if i call that channel, then * answers the call and plays the menu - but my choice is not processed... the call simply times out without connecting to the phones - what am i doing wrong? |
| 04:15.18 | SwK | if it sounds too loud turn it down |
| 04:15.27 | SwK | if not loud enuff turn it up |
| 04:15.31 | Drew | [mainmenu] |
| 04:15.31 | Drew | exten => s,1,Answer |
| 04:15.31 | Drew | exten => s,2,Wait(2) |
| 04:15.31 | Drew | exten => s,3,Background(tt-monkeysintro) |
| 04:15.31 | Drew | exten => 1,1,Dial(SIP/phone1) |
| 04:15.32 | Drew | exten => 2,1,Dial(SIP/phone2) |
| 04:15.39 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 04:15.54 | budmang | Defraz teliax hook up directly to asterisk with the 24/95 plan? |
| 04:16.06 | Defraz | yea |
| 04:16.13 | Defraz | that is what I have two simo. connections |
| 04:16.18 | SPoon_TSX | Swk: I got the rx right. Where I can hear the caller. But no matter how I adjust the tx, I get the echo. The higher number the worest. |
| 04:17.47 | budmang | can i pm you |
| 04:17.47 | SwK | so turn down the tx gai |
| 04:17.48 | budmang | its blocked |
| 04:17.55 | Drew | any ideas on why my menu isnt working? |
| 04:18.09 | Defraz | so they don't deliver CW? |
| 04:18.24 | *** join/#asterisk aster (i=EvilInLo@ws1.bratatex.melsa.net.id) |
| 04:18.27 | znoG | SPoon_TSX: what sort of phone are you using to connect to * ? |
| 04:18.33 | budmang | helpme channel |
| 04:18.40 | SPoon_TSX | Swk: My rx is 8 now, and I hear the caller clearly with some statics noise. From your experties, would you consider that is too loud? |
| 04:18.49 | SPoon_TSX | znoG: PolyCOM 501. |
| 04:18.53 | znoG | right |
| 04:18.57 | aster | hello ... |
| 04:19.03 | znoG | do you have echo cancellation on, etc? |
| 04:19.10 | SwK | spoon_tx: not nesecarily |
| 04:19.17 | znoG | echocancel=yes |
| 04:19.17 | znoG | echocancelwhenbridged=yes |
| 04:19.17 | znoG | echotraining=yes |
| 04:19.26 | SwK | spoon_tx: call yourself on your cellphone and see how that works |
| 04:19.34 | SPoon_TSX | znoG: yes. I got echocancel=64. |
| 04:19.50 | aster | where can i put new user extentions without amp |
| 04:19.55 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 04:20.00 | aster | where can i put new user extentions without amp |
| 04:20.05 | znoG | SPoon_TSX: no idea, i suggest reading all the echo cancel stuff on voip-info.org |
| 04:20.28 | znoG | SPoon_TSX: also there's a utility in the asterisk source that helps with determining your {rx,tx}gain settings |
| 04:20.31 | znoG | ztmonitor |
| 04:20.44 | znoG | actually it helps to see if you need to decrease and/or increase |
| 04:21.13 | SPoon_TSX | znoG: I got that tools, but I just don't know how that things work. |
| 04:21.25 | budmang | Defraz where did you get help with connecting with it? |
| 04:22.16 | SPoon_TSX | Swk: I called my cell, I can hear myself clearly, but I can also hear some minder echo. Any idea? |
| 04:22.23 | *** join/#asterisk Muckl (n=yo@p54BED3FB.dip.t-dialin.net) |
| 04:22.28 | budmang | IM about to ditch my vonage for it |
| 04:22.49 | budmang | and try to integrate to my dedicated server in a colo asterisk server |
| 04:23.08 | aster | where can i put new user extentions without amp |
| 04:23.25 | Drew | i have a problem with creating menus in the extensions.comf: - i made a menu so incoming calls could choose which phone they want to ring - if i call that channel, then * answers the call and plays the menu - but my choice is not processed... the call simply times out without connecting to the phones - what am i doing wrong? |
| 04:23.26 | Drew | [menu] |
| 04:23.26 | Drew | exten => s,1,Answer |
| 04:23.26 | Drew | exten => s,2,Wait(2) |
| 04:23.26 | Drew | exten => s,3,Background(tt-monkeysintro) |
| 04:23.28 | Drew | exten => 1,1,Dial(SIP/phone1) |
| 04:23.30 | Drew | exten => 2,1,Dial(SIP/phone2) |
| 04:23.34 | SwK | spoon_tsx: have you tried leaving the rxgain alone and turning down the txgain |
| 04:23.45 | fdask | Drew: any debug output of the call? |
| 04:23.47 | SwK | drew: thats not a way to get an answer, thats a good way to get kicked |
| 04:23.51 | joaovianna | Hi guys! I have a question. Can I have a particular [context] for a particular incoming sip call based on ip or domain ? |
| 04:23.54 | fdask | you try upping the verbose and checking for clues? |
| 04:24.07 | SwK | joaovianna yes |
| 04:24.45 | Drew | fdask - you mean what "SIP DEBUG" says? |
| 04:24.59 | joaovianna | SwK: Thanks... How ? I'm trying to setup a type=user (incoming) in sip with a context but it is not working. |
| 04:25.10 | Drew | SwK - sorry - im new to this, and slightly frustrated, dont take it personally |
| 04:25.41 | SwK | joavianna is there a matching peer or friend for the same IP address? |
| 04:26.11 | Defraz | Just reading some forms on caller ID |
| 04:26.28 | fdask | Drew: asterisk log files, or console output |
| 04:26.36 | SwK | drew: I'm not |
| 04:26.51 | SwK | drew: just letting you know before an op gets cranky |
| 04:26.57 | SwK | ;) |
| 04:27.05 | joaovianna | Swk: No. Now I'm confused. I want to receive a sip call from my other * (sip), and I want to have it going to a different context than default. |
| 04:27.11 | SPoon_TSX | Swk: If I set rx=0, tx=-1. The echo is almost gone. Should I go lower the tx to completely get rip of the tx on this settings? |
| 04:27.48 | SwK | its possible |
| 04:27.58 | SwK | spoon: going up is not always the answer |
| 04:28.01 | Drew | console: it answers the call normally, shows that its playing the tt-monkeysintro sound file, then outputs "[date} WARNING[3239]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'menu' |
| 04:28.12 | SwK | turning up the gaines to high can result in more echo not less echo |
| 04:28.14 | Drew | SwK - thx |
| 04:28.19 | SPoon_TSX | Swk: Oops.. when I use -2, the echo come worest again. |
| 04:28.26 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:28.26 | znoG | Drew: tried: exten => s,4,WaitForExten ? |
| 04:28.41 | Drew | nope - ill try that.. |
| 04:28.48 | znoG | Drew: i mean WaitExten |
| 04:28.51 | znoG | Drew: not WaitForExten |
| 04:28.56 | SwK | you get echo on the lines when using a pots phone? |
| 04:29.08 | SPoon_TSX | Swk: Yes. |
| 04:29.41 | SwK | so you do get echo without asterisk in the loop? |
| 04:29.42 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 04:29.53 | SPoon_TSX | Swk: Yes. |
| 04:30.19 | joaovianna | Swk: Can you explain "matching peer" ? The call is coming from my * box. Both with static IP. |
| 04:30.25 | SwK | spoon_tsx: ok stop screwing with zapata.conf and call the telco and cuss them they have a bad connection sonewhere heh |
| 04:30.45 | znoG | SPoon_TSX: if you get echo without asterisk in the loop, why are you trying to fix it within asterisk? |
| 04:30.53 | SwK | exactly |
| 04:31.02 | SwK | thats something the telco has to dix |
| 04:31.04 | SwK | err fix |
| 04:31.06 | znoG | right |
| 04:31.07 | SPoon_TSX | Oh, no. I mean I only get echo WITH Asterisk. |
| 04:31.21 | znoG | ah. |
| 04:31.30 | znoG | SPoon_TSX: you should get support from Digium anyway if its a TDM400 |
| 04:31.43 | znoG | so you could try them |
| 04:31.54 | znoG | or read all the docs on voip-info regarding echo cancellation |
| 04:32.05 | znoG | SPoon_TSX: another reason for echo could be that the TDM400 is sharing an IRQ with some other card |
| 04:32.10 | znoG | cat /proc/interrupts |
| 04:32.13 | SPoon_TSX | znoG: I tried them, they told me excatly the same thing as what you told me. |
| 04:32.14 | SwK | joaovianna: want the simple fix? |
| 04:32.36 | joaovianna | SwK: IAX ? |
| 04:33.06 | SPoon_TSX | After I typed the cat command, what should I look for? |
| 04:33.16 | joaovianna | SwK: Yes please. |
| 04:33.17 | SwK | joaovinna: set up a type=friend, host=ip.of.other.box, insecure=very, permit=ip.of.other.box/255.255.255.255, deny=0.0.0.0/0.0.0.0, context=whatever |
| 04:33.19 | znoG | SPoon_TSX: mind you I just purchased a TDM400 myself and I left the office with echo problems, just like you're having, but I haven't even begun playing with rx/tx gain settings |
| 04:33.34 | znoG | SPoon_TSX: do you have the driver for the TDM on its own line? |
| 04:33.46 | znoG | SPoon_TSX: or is there like: <module>, <driver for tdm> |
| 04:34.02 | SPoon_TSX | It said: 3: 13026448 XT-PIC SysKonnect SK-98xx, wctdm |
| 04:34.11 | znoG | looks like its sharing an interrupt |
| 04:34.13 | SwK | joaovinna: sip peer on each box, then dial(sip/name_of_peer/${EXTEN}) and move on |
| 04:34.17 | znoG | im not saying thats the problem, but it may be |
| 04:34.35 | SwK | that could be a big problem |
| 04:34.38 | znoG | Drew: did it work? |
| 04:34.41 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 04:34.44 | SwK | and make sure you arent getting missed interupts |
| 04:34.45 | SPoon_TSX | znoG: but what is SysKonnect SK-98xx? |
| 04:34.51 | znoG | that i can't answer |
| 04:34.56 | Drew | znoG - i tryed adding WaitExten(5) - doesnt change anything |
| 04:34.57 | znoG | sounds like a com port or something on irq 3 |
| 04:35.13 | SwK | the SysKonnect is probably a network card |
| 04:35.21 | znoG | ah then.. yes.. could be a problem |
| 04:35.29 | Drew | i call the channel, wait for the message, press i.e. 1 - but nothing happens - simply times out |
| 04:35.36 | Dr-Linux | http://pastebin.com/521821 |
| 04:35.40 | fdask | is that keypress registering in asterisk |
| 04:35.41 | Dr-Linux | SwK |
| 04:35.47 | znoG | Drew: www.pastebin.com - show your config |
| 04:35.49 | SPoon_TSX | I got the onboard network card. Any idea I can get its own IRQ? |
| 04:35.53 | Drew | there is no console output on the keypress |
| 04:35.57 | znoG | Drew: im thinking the problem is your DTMF setup |
| 04:35.57 | SwK | drew: whats is the call coming in on SIP |
| 04:36.03 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 04:36.20 | Drew | that of * or that of the hardline i am calling from? |
| 04:36.22 | joaovianna | SwK: Worked ! It is not suposed to be "user" instead "friend" ? Thanks. |
| 04:36.29 | znoG | Drew: asterisk is not receiving the tones. try dtmfmode=auto in the sip/iax.conf relevant section for the polycom |
| 04:37.17 | SwK | joaovianna: friend == user + peer |
| 04:37.34 | SwK | so now with those you can send calls back and forth between the 2 asterisk boxes with no problem |
| 04:37.35 | znoG | Drew: or was it SPoon_TSX that has the polycom, anyway whatever user you're using to dial the menu with, give them dtmfmode=auto and see if that works (reload asterisk after that, of course) |
| 04:37.47 | SPoon_TSX | Yes, Swk, you right. It is a network card. Is there anyway to make wctdm use some other IRQ? |
| 04:38.03 | Drew | thanks that did the trick :D |
| 04:38.07 | SwK | if you wish to restrict it to one way change them to type=user and type=peer on the appropriate sides |
| 04:38.08 | ravenpi | What kind of forwarding should I use for the Polycoms? When I go with the phone's own forwarding (with canreinvite=no in Asterisk), I get about 10 seconds of silence (no MoH) -- the transferee thinks they've been hung up on. |
| 04:38.28 | znoG | SPoon_TSX: play with the BIOS settings for the IRQs |
| 04:38.37 | znoG | SPoon_TSX: i think switching PCI slots sometimes helps |
| 04:38.46 | Drew | i had used some example configuration that had dmtfmode set to info... |
| 04:38.52 | SPoon_TSX | any other way? |
| 04:39.00 | SPoon_TSX | Since I am not in front of my machine now. |
| 04:39.09 | SwK | spoon_tsx: use a different nic? heh |
| 04:39.48 | SwK | spoon_tsx: at this point you almost must have physical access or an IP-KVM to make bios adjustments OR you have to physically move the card to a different slow |
| 04:39.52 | SwK | err slot |
| 04:40.51 | SPoon_TSX | Damn, is there no command line to specify the IRQ for the driver in Linux? |
| 04:41.15 | SwK | damn theres like 12 people haveing * -> * call issues and a different 12 all having echo issues tonight |
| 04:41.16 | *** join/#asterisk FuriousGeorg1 (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 04:41.46 | SwK | and all the echo issues are like TDM400s hahaha |
| 04:41.55 | znoG | SPoon_TSX: not that i know of |
| 04:41.57 | znoG | anyway, sleep time |
| 04:42.00 | znoG | good luck with it! |
| 04:42.26 | aster | hello i'm a newbie here so... my question is i want to add new extentions in asterisk |
| 04:42.30 | aster | where can i do that ? |
| 04:42.41 | znoG | not only a newbie, but a lazy one at that :) |
| 04:42.45 | *** join/#asterisk VirtualUK (n=newkinet@shell2.sea5.speakeasy.net) |
| 04:42.48 | znoG | http://www.voip-info.org has all the info you need |
| 04:42.56 | aster | :D |
| 04:43.13 | VirtualUK | anyone know how I can change the IRQ of a PCI card? It's got an unsually high IRQ # (37) at the moment and I think it's the root of my problems |
| 04:43.27 | drumkilla | aster: specifically, look at extensions.conf :) |
| 04:43.45 | SwK | vitualuk: thats not a real IRQ... thats a mapped IRQ via APIC |
| 04:44.32 | SPoon_TSX | Swk: Damn! But thanks for all of your help. |
| 04:45.21 | VirtualUK | SwK: got a strange problem, my card shows up with lspci but modprobe wcfxo fails with a ZT_CHANCONFIG error :( |
| 04:45.45 | Psykick | quick question |
| 04:46.09 | Psykick | is anyone running a stable asterisk system on linux that can handle large amounts of concurrent calls |
| 04:46.14 | Psykick | if so please pm me |
| 04:46.53 | budmang | whats a good instant voip provider that works with asterisk? |
| 04:47.09 | SwK | zt_chanconfig error? |
| 04:47.11 | SwK | hah |
| 04:47.31 | VirtualUK | yup |
| 04:47.32 | *** join/#asterisk linlin (n=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 04:47.38 | VirtualUK | driving me crazy |
| 04:47.44 | SwK | whats autorunning ztconfig for you? |
| 04:47.50 | SwK | err ztcfg |
| 04:47.56 | budmang | whats a good instant voip provider that works with asterisk? |
| 04:48.27 | SwK | fix your /etc/zaptel.conf or make it so ztcfg does not autprun in the appropriate place for your distro |
| 04:48.35 | VirtualUK | SwK : does the same thing |
| 04:48.40 | SwK | budmang: asterlink |
| 04:48.48 | SwK | lsmod show it as loaded? |
| 04:49.07 | znoG | heh, would be nice if one bothers to help somebody else, that person also bothers to thank the person who helped him/her, but I guess that's asking a bit too much eh. |
| 04:49.19 | VirtualUK | SwK : yup, but asterisk chokes on it |
| 04:49.31 | SwK | what does ztcfg -vvvvvvvvvv |
| 04:49.33 | SwK | say? |
| 04:49.49 | ke4qqq | anyone know of a reference online for a dialplan that allows you to check vm after being forwarded to vm (ie, press * or # during the playing of the recording (busy or unavailable) and be given the auth prompt?) |
| 04:50.04 | VirtualUK | vault:~ # ztcfg -vvvvvvvvvvvvv |
| 04:50.04 | VirtualUK | Zaptel Configuration |
| 04:50.05 | VirtualUK | ====================== |
| 04:50.05 | VirtualUK | Channel map: |
| 04:50.05 | VirtualUK | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 04:50.05 | VirtualUK | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 04:50.07 | VirtualUK | 2 channels configured. |
| 04:50.09 | VirtualUK | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 04:50.47 | SwK | is that the correct channel configs for your card? |
| 04:51.06 | VirtualUK | according to the docs yup, it's a X100P clone |
| 04:51.15 | SwK | 1 card or 2 |
| 04:51.23 | NewSole | VirtualUK you have power cable pluged into card |
| 04:51.23 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 04:51.28 | ravenpi | Is it the FXO that dies horribly if the power isn't plugged in? |
| 04:51.28 | VirtualUK | 1 card, one fxo port, one fxs port |
| 04:51.32 | ravenpi | Wups -- you beat me to it. |
| 04:51.39 | SwK | then its not a X100P cloan |
| 04:51.47 | SwK | X100P clones are only 1 FXO port |
| 04:51.51 | SwK | they are modems |
| 04:52.26 | SwK | they all have 2 ports on the back but look at them and you'll see one has a pictogram or is labeled for the wall the other for a phone |
| 04:52.40 | VirtualUK | right |
| 04:52.43 | VirtualUK | one is a fxs port |
| 04:52.45 | VirtualUK | one is a fxo port |
| 04:52.51 | VirtualUK | like this one http://cgi.ebay.com/Clone-of-Digium-X100P-X101P-Wildcard-for-Asterisk-PBX_W0QQitemZ5856427611QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 04:52.59 | budmang | swk: asterlink where on there site i see a link that goes no where |
| 04:53.01 | znoG | err no, one is a fxo port, the other is just a phone port, but not FXS |
| 04:53.10 | znoG | FXS means you can configure it in Asterisk as a regular phone, and this is not the case |
| 04:53.13 | SwK | ITS A FUCKING MODEM |
| 04:53.28 | VirtualUK | lol...wow, temper temper |
| 04:53.29 | SwK | IT IS ONLY AN FXO CARD there is NOT FXS Slick on that card |
| 04:53.34 | SwK | the 2 jacks are for passthru |
| 04:53.44 | SwK | sorry |
| 04:53.50 | Katty | mew. |
| 04:53.53 | VirtualUK | lol |
| 04:54.10 | SwK | and if you paid $30 bucks for it you got ripped off |
| 04:54.17 | VirtualUK | nope, I paid $8 |
| 04:54.20 | budmang | SWK asterlink doesnt seem to work |
| 04:54.39 | SwK | asterlink works fine |
| 04:54.45 | NewSole | VirtualUK you have power cable pluged into card |
| 04:55.00 | SwK | there is no powercable for an X100P |
| 04:55.02 | VirtualUK | NewSole : what are you on about? it's a PCI card |
| 04:55.05 | VirtualUK | lol |
| 04:55.06 | Katty | SwK: :< |
| 04:55.07 | SwK | its just a cheap ass windows modem |
| 04:55.10 | VirtualUK | right |
| 04:55.11 | fugitivo | powercable for the x100p, lol |
| 04:55.18 | NewSole | sorry thought it was tdm crap |
| 04:55.29 | SwK | hey katty |
| 04:55.41 | Katty | SwK: :> |
| 04:55.44 | file | budmang: what's not working about it? |
| 04:55.44 | *** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net) |
| 04:55.50 | VirtualUK | it's a tiger jet 3xx card, but I'm buggered if it'll work at all |
| 04:56.03 | budmang | the site |
| 04:56.24 | budmang | <PROTECTED> |
| 04:56.24 | budmang | PB1X |
| 04:56.24 | budmang | Asterlink Linux distribution built specifically for Asterisk |
| 04:56.24 | budmang | <PROTECTED> |
| 04:56.24 | budmang | COMING SOON |
| 04:56.48 | *** join/#asterisk pimpwell (n=pimpwell@ool-44c768ec.dyn.optonline.net) |
| 04:57.06 | SwK | budmang: thats their linux distro... look at their voip services |
| 04:57.25 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:57.31 | SwK | virtualuk: nuk the config for the fxs port, that card doesnt have one |
| 04:57.40 | SwK | and remember that fxo ports get fxs signalling |
| 04:57.43 | budmang | I want to sign up and play now is it instant? |
| 04:57.49 | file | budmang: yes... |
| 04:57.51 | fdask | so asterlink provide voip call termination? |
| 04:57.53 | Katty | file! |
| 04:57.57 | fdask | couldnt really understand that site |
| 04:57.58 | file | Katty! |
| 04:58.17 | VirtualUK | SwK : thanks I'll give it a try |
| 04:58.54 | file | it's getting far too late... |
| 04:58.58 | file | I can't spell ping right |
| 04:58.59 | budmang | i just need 2 connections @ the same time for about 30/month |
| 04:59.01 | Katty | file: naptime for you mister. |
| 04:59.10 | file | budmang: we do per minute |
| 04:59.16 | Katty | file: (= |
| 04:59.19 | budmang | file do you have a channel |
| 04:59.21 | budmang | we can talk in |
| 04:59.22 | budmang | i cant pm |
| 04:59.24 | file | #asterlink |
| 04:59.29 | fugitivo | 404 for the asterlink linux distro |
| 04:59.31 | rajiv | great use of asterisk http://jabphone.com/ |
| 05:00.05 | VirtualUK | SwK : no worries ;) |
| 05:00.42 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 05:03.38 | dmz | rajiv, that is cool... |
| 05:04.19 | rajiv | asterisk may be but they probably wrote their app as AGI or * applications, in which case they do not need to release |
| 05:04.35 | rajiv | actually, they do not need to release in any case bc they are not shipping any binaries |
| 05:05.48 | fugitivo | isn't that just a service? |
| 05:05.53 | rajiv | yes |
| 05:05.58 | fugitivo | so? |
| 05:06.03 | fugitivo | who cares about code? |
| 05:07.07 | rob0 | In Soviet Russia code cares about YOU! |
| 05:07.34 | Katty | yay! |
| 05:10.04 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 05:11.11 | dmz | i do, i want to set my own up, why pay to call myself :) |
| 05:13.26 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 05:13.37 | SwK | katty: why you gonna shoot file with your bow? |
| 05:14.12 | Katty | SwK: who would bring me muffins in the morning then? |
| 05:16.09 | SwK | katty: i'll set up a cronjob to fax tehm to you |
| 05:16.16 | Katty | :< |
| 05:16.33 | file | FAX |
| 05:16.34 | file | WHAT |
| 05:16.34 | file | NO |
| 05:16.42 | file | GAH |
| 05:16.56 | fdask | in my dialplan, when i do a Dial command, the number gets dialed ok, but asterisk dies shortly after it thinks its bridged the lines |
| 05:17.11 | Katty | less faxing, more hugging. |
| 05:17.12 | fdask | the dialed number keeps ringing, and if i answer, nothing is there |
| 05:17.15 | SwK | hahaha |
| 05:17.28 | fdask | asterisk: vpbapi.cpp:3403: int vpb_bridge(int, int, int, int): Assertion `0' failed. |
| 05:17.35 | SwK | make war not love! |
| 05:17.36 | fdask | looks like a problem with my cards driver |
| 05:17.38 | fdask | :( |
| 05:17.41 | SwK | errrrrr.... Make Love not War! |
| 05:17.50 | Katty | make config, not war. |
| 05:18.01 | SwK | make install clean |
| 05:18.11 | file | or... |
| 05:18.12 | file | jbot: sex? |
| 05:18.14 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 05:18.19 | Katty | hottt. |
| 05:19.36 | SwK | hah |
| 05:19.41 | SwK | "or super extractor" |
| 05:20.11 | zamsler_ | lol |
| 05:20.36 | zamsler | hooah |
| 05:21.32 | litage | why does asterisk ship with such an old version of openh323? |
| 05:22.00 | *** join/#asterisk torrent_corp (n=alexis@toronto-HSE-ppp4331187.sympatico.ca) |
| 05:23.16 | fdask | can i get asterisk to perform a dial operation from the cli? |
| 05:23.20 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 05:23.26 | torrent_corp | hello ... i have a problem with my directory ... even if i record names in the voicemail the dorectory is still spelling the name ... anyone know why and how i can fix it ??? |
| 05:23.31 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 05:24.44 | SwK | fdask: try a .call file or a manager command |
| 05:25.02 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com) |
| 05:25.04 | fdask | ok |
| 05:25.44 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 05:26.21 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 05:27.46 | torrent_corp | no one have an idea? |
| 05:28.24 | torrent_corp | or only a little clue :) |
| 05:28.50 | *** join/#asterisk alphaque (n=alphaque@60.48.153.162) |
| 05:29.26 | Corydon76-home | torrent_corp: probably a permissions problem |
| 05:33.00 | torrent_corp | no ... and the voicemail play the name |
| 05:36.52 | fdask | asterisk crashes when i execute a dial() statement in my dialplan. looks like its attempting to bridge the two lines before the second has answered |
| 05:36.54 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.15.43.telnor.net) |
| 05:36.54 | fdask | is that normal? |
| 05:37.02 | fdask | like it attempting to bridge while the dial() is still dialing? |
| 05:38.12 | zamsler | hmm |
| 05:38.30 | zamsler | it happen all the time... |
| 05:38.34 | zamsler | Kidding. |
| 05:38.50 | zamsler | what version are you running? |
| 05:39.17 | zamsler | does it just segfault? |
| 05:39.21 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 05:40.08 | *** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 05:42.58 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 05:43.44 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:45.09 | knight_ | everyone beware, FVDSL (FordVoice) is a scam |
| 05:45.24 | *** join/#asterisk Drew (n=foo@zux221-065-169.adsl.green.ch) |
| 05:46.07 | torrent_corp | ok i just find my problem ... when an extention get a new voicemail asterisk record it in /var/spool/asterisk/... but when i record an unavailable message or a name asterisk put it in /var/lib/asterisk/sounds/voicemail/... anyone know why? |
| 05:48.23 | torrent_corp | its my 5th asterisk server and i never saw that before |
| 05:49.00 | *** join/#asterisk scon_ (n=scon@dslb-084-057-014-201.pools.arcor-ip.net) |
| 05:55.27 | zamsler | knight_, agreed they are too expensive anyway.. |
| 05:56.32 | *** join/#asterisk j0dan (n=dan@bb219-75-102-245.singnet.com.sg) |
| 05:56.47 | zamsler | knight_, do u need termination? |
| 05:59.50 | *** join/#asterisk clive- (n=pirch@dsl-165-151-101.telkomadsl.co.za) |
| 06:00.08 | knight_ | zamsler, not at this moment... but i did some work for them, so i know very clearly how full of crap they are |
| 06:00.17 | zamsler | oh ok. |
| 06:00.19 | knight_ | they scammed me out of $6000 worth of consulting |
| 06:00.37 | knight_ | not to mention they just piggy back on sellvoip |
| 06:00.44 | knight_ | and mark it up several cents |
| 06:00.44 | *** join/#asterisk Cool_One (n=bclinton@h84.65.255.206.cable.htsp.cablelynx.com) |
| 06:00.50 | zamsler | LOL |
| 06:00.50 | Cool_One | HELP |
| 06:00.52 | zamsler | hahahaha |
| 06:01.06 | knight_ | they have no true termination |
| 06:01.10 | Cool_One | can anyone tellme what this error means |
| 06:01.14 | Cool_One | <PROTECTED> |
| 06:01.15 | Cool_One | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_cust_config_register |
| 06:01.15 | Cool_One | [root@pbx asterisk]# Warning, flexibel rate not heavily tested! |
| 06:01.15 | Cool_One | Ouch ... error while writing audio data: : Broken pipe |
| 06:01.35 | Drew | how do i turn off the native bridging of sip calls? ie so that a sip call is relayed through * instead of being transported directly between the 2 peers? |
| 06:01.38 | zamsler | knight_, but what is true termination? |
| 06:01.39 | clive- | dos anyone use mondo/mindi here ? |
| 06:01.48 | knight_ | zamsler, indeed |
| 06:01.56 | zamsler | lol |
| 06:02.02 | zamsler | I provide termination.. |
| 06:02.12 | knight_ | i dont consider buying some minutes from sellvoip and marking it up to be a quality termination company |
| 06:02.25 | zamsler | and I use sellvoip, and sixtel, and voxee, etc. |
| 06:02.33 | zamsler | I have never had any quality issues. |
| 06:02.52 | zamsler | I also use XO, but I do LCR, so whoever is the best.... |
| 06:03.17 | knight_ | yep |
| 06:03.19 | knight_ | which is fine |
| 06:03.45 | zamsler | I know some of the CLECS that do voip, just shuffle via LCR. |
| 06:04.01 | zamsler | once it is voip, it doesn;t matter how it terminates.. Long as there is QOS |
| 06:04.01 | knight_ | but he was using a P3 on a dsl line, $50 in credits to sellvoip, and saying he had all sorts of piered termination |
| 06:04.06 | clive- | zamsler is this for SIP of for iax2? |
| 06:04.13 | zamsler | I do SIP. |
| 06:04.15 | knight_ | oh yeah, he claims he is now a CLEC in Ohio |
| 06:04.19 | Cool_One | can someone lend a brain? |
| 06:04.29 | zamsler | lol |
| 06:04.35 | zamsler | it is possible |
| 06:04.41 | knight_ | no i know he isnt |
| 06:04.44 | zamsler | in Montana costs like $20 to be a CLEC |
| 06:04.47 | knight_ | he's a broke con artist |
| 06:05.17 | zamsler | lots of people are. |
| 06:05.24 | zamsler | ;) |
| 06:05.32 | zamsler | some of us actually try. |
| 06:05.45 | zamsler | ant those idiots give us small guys a bad name. |
| 06:06.15 | *** join/#asterisk osirus (n=jbrewer@adsl-68-127-117-213.dsl.frsn02.pacbell.net) |
| 06:06.17 | reza | ok, bought a new system - tdm400 shows up in lspci |
| 06:06.21 | knight_ | heh |
| 06:06.36 | zamsler | knight_, what kinda consulting did you do for them? |
| 06:06.49 | reza | i forgot what docs i was using before - it told me to insmod a module to test - does anyone know the module name or have a link to those docs? |
| 06:07.21 | zamsler | wct4xxp |
| 06:07.21 | knight_ | zamsler, customize astbill scripting, rate card conversion scripts, did handling, etc |
| 06:07.32 | zamsler | reza, wct4xxp |
| 06:07.46 | zamsler | knight_, Cool! I was going to use astbill |
| 06:07.58 | zamsler | but it takes up too much resources. |
| 06:08.01 | osirus | whatup |
| 06:08.02 | knight_ | zamsler, yeah astbill is pretty slick... still immature, but useful |
| 06:08.08 | knight_ | yep |
| 06:08.20 | zamsler | I wrote my LCR and billing in C |
| 06:08.44 | reza | zamsler - thanks |
| 06:08.50 | zamsler | After I write a customer front end then I will start advertising. |
| 06:08.58 | zamsler | I need to do up a website also. |
| 06:09.03 | zamsler | reza, NP! |
| 06:09.25 | reza | insmod: error inserting './2.6.14-1.1656_FC4/misc/wct4xxp.ko': -1 Unknown symbol in module |
| 06:09.25 | zamsler | everyone wants a website. |
| 06:09.31 | zamsler | hmm |
| 06:09.34 | reza | grr |
| 06:09.41 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 06:10.02 | osirus | zamsler: mambo will get you up and running prettty quick |
| 06:10.07 | zamsler | sounds like you have some issues... Google that message.. I read something about it. |
| 06:10.26 | zamsler | osirus, :) of course.. Great minds think alike. |
| 06:10.45 | osirus | its the best thing since sliced ham |
| 06:10.53 | zamsler | Yup |
| 06:11.07 | zamsler | I tried that joomala also. |
| 06:11.11 | zamsler | same difference. |
| 06:11.13 | osirus | yeah |
| 06:11.15 | knight_ | joomla is the new mambo |
| 06:11.29 | zamsler | content management.. |
| 06:11.33 | zamsler | blah blah |
| 06:11.44 | zamsler | I haven't found any that are better. |
| 06:11.47 | osirus | i use it for my blog |
| 06:11.52 | osirus | sigh |
| 06:12.20 | zamsler | LOL.. |
| 06:12.24 | *** join/#asterisk john_ (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 06:12.33 | zamsler | someone wanted to charge $200 for a mambo skin. |
| 06:12.58 | osirus | you dont even have to really |
| 06:13.02 | zamsler | lol |
| 06:13.03 | zamsler | nope. |
| 06:13.06 | osirus | theresa shitload of free ones |
| 06:13.13 | zamsler | yup all over. |
| 06:13.23 | zamsler | but I am a php programmer. |
| 06:13.29 | zamsler | so i did that in my sleep. |
| 06:13.32 | osirus | ah |
| 06:13.42 | zamsler | and with some macromedia |
| 06:13.45 | zamsler | ;) |
| 06:13.54 | zamsler | I used to code with vim... |
| 06:14.04 | osirus | baah |
| 06:14.07 | osirus | im an og |
| 06:14.11 | zamsler | Zend and Macromedia makes things so much easier. |
| 06:14.13 | osirus | i used to code with edlin |
| 06:14.16 | zamsler | lol |
| 06:14.25 | zamsler | and you charge by the line.. |
| 06:14.35 | osirus | haha |
| 06:14.37 | zamsler | I might have a y2k project that I never finished.. |
| 06:14.39 | zamsler | ;) |
| 06:15.01 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
| 06:15.14 | osirus | haha |
| 06:15.23 | c4r3x | hello |
| 06:15.34 | osirus | thats funny, hows that for a trip down memory lane |
| 06:15.40 | osirus | edlin |
| 06:15.43 | osirus | ffff |
| 06:15.45 | c4r3x | any body here have been configure asterisk on openbsd box??? |
| 06:15.50 | zamsler | ewwww |
| 06:15.52 | zamsler | bsd.. |
| 06:15.57 | zamsler | ever heard of... |
| 06:15.59 | zamsler | DEBIAN? |
| 06:16.16 | osirus | theres lots of stuff in the wiki about asterisk and freebsd |
| 06:16.25 | osirus | ithink |
| 06:16.29 | zamsler | yeah there is. |
| 06:16.35 | c4r3x | osirus --> i mean openbsd |
| 06:16.43 | zamsler | but things run so much better on debian. |
| 06:16.45 | c4r3x | it's litle diffrent |
| 06:17.00 | zamsler | grab a hammer and pound it out,. |
| 06:17.28 | *** part/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
| 06:18.22 | osirus | c4r3x: why openbsd? |
| 06:19.05 | zamsler | openbsd is very secure. |
| 06:19.14 | osirus | sigh |
| 06:19.15 | osirus | here we go |
| 06:19.24 | zamsler | lol |
| 06:19.30 | zamsler | I still like debian. |
| 06:19.31 | zamsler | ;) |
| 06:20.03 | osirus | doesnt digium suggest you put it on a debian distro anyway |
| 06:20.16 | zamsler | i think so. |
| 06:20.17 | zamsler | lol |
| 06:20.17 | osirus | or something |
| 06:20.19 | osirus | what is it |
| 06:20.58 | Drew | i have a problem with native bridging of sip calls - how do i deactivate it? ie. so that all calls are relayed through the *-box? |
| 06:21.11 | osirus | its like xorcom |
| 06:21.14 | wasim | reinvite=no |
| 06:21.15 | zamsler | canreinvite=no |
| 06:21.34 | Drew | yes i added that to my sip.conf - but it doesnt seem to have any effect |
| 06:21.41 | osirus | sip reload? |
| 06:21.45 | Drew | done that |
| 06:22.00 | Drew | i eaven restarted the hole asterisk |
| 06:22.22 | zamsler | which one did you add? |
| 06:22.42 | Drew | canreinvite=no |
| 06:22.51 | zamsler | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 06:23.08 | zamsler | the only thing we can do is shrug and point at the manual.. |
| 06:23.17 | zamsler | i mean it just works. |
| 06:23.17 | Drew | ... |
| 06:23.20 | zamsler | :( |
| 06:25.46 | rob0 | OpenBSD is only really secure out-of-the-box. You set up services on it, you open potential vulnerabilities. It's good that they put so much effort into being secure out of the box, but what does that mean to most people? Little to nothing. |
| 06:25.57 | Drew | do i add these options to the [general] context or to the contexts of each channel/phone/sipaccount? |
| 06:26.27 | wasim | either |
| 06:27.34 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 06:27.39 | kuku5 | which port does vpn use ? |
| 06:29.24 | zamsler | umm.. what does that have to do with asterisk? |
| 06:29.46 | zamsler | VPNs can use what ever port you tell them to. |
| 06:30.42 | kuku5 | ipsec |
| 06:30.49 | pifiu-laptop | later guys talk to you guys tomorrow |
| 06:30.57 | zamsler | later |
| 06:33.06 | *** join/#asterisk sac|h0p|werk|afk (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
| 06:34.18 | *** join/#asterisk Assid (n=assid@203.115.64.10) |
| 06:34.33 | Assid | woohoo |
| 06:35.08 | *** join/#asterisk NghtLght (n=ircproxy@202.179.3.171) |
| 06:35.12 | kuku5 | port 1723 |
| 06:35.13 | Drew | it just isnt working - it always tries to natvie brige the sip calls |
| 06:38.46 | Drew | does sip only use port 5060? |
| 06:38.52 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 06:39.19 | zamsler | no |
| 06:39.48 | Drew | what range does it use? |
| 06:39.58 | zamsler | kuku5, where u from? |
| 06:40.15 | zamsler | Drew, what ever you tell it to.. Did u RTFM?? |
| 06:42.13 | *** join/#asterisk riksta (n=rick@62.6.163.81) [NETSPLIT VICTIM] |
| 06:42.13 | *** join/#asterisk kll (i=kll@insomnia.juniks.net) [NETSPLIT VICTIM] |
| 06:42.13 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) [NETSPLIT VICTIM] |
| 06:43.44 | Drew | sorry zamsler - ive spent the whole night R'ingTFM - and im kinda pissed at the moment because nothing is working |
| 06:44.06 | zamsler | this stuff just works out of the box. |
| 06:44.14 | zamsler | never had an issue. |
| 06:44.19 | Z-Knight | anyone here developing a JAVA based front end to Asterisk? |
| 06:44.19 | c4r3x | hahahahahha |
| 06:44.38 | c4r3x | asterisk is work fine in openbsd 3.8 |
| 06:44.56 | tainted- | anyone have a problem where after installing mpg123 all audio fails to work? |
| 06:45.05 | zamsler | Drew, i dunno what to tell you.. |
| 06:45.20 | Drew | zamsler and c4r3x obiously you are lucky... |
| 06:45.34 | zamsler | the example files are very well documented. |
| 06:45.41 | Drew | but the options i am setting arent takeing any effect |
| 06:45.50 | zamsler | of ffs. |
| 06:46.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:46.14 | Drew | ffs? |
| 06:46.29 | zamsler | for f!@#s sakes |
| 06:46.42 | mut | anyone know how i can get my registration code for my g729 codec |
| 06:46.46 | mut | from my current install |
| 06:46.53 | mut | cause i don't have the email with it anymore |
| 06:47.24 | zamsler | Drew, sorry.. |
| 06:47.38 | zamsler | there are like 350 people in here that don't have an issue. |
| 06:47.54 | Drew | i understand your PoV... |
| 06:47.55 | zamsler | and 51 that have never istalled asterisk. |
| 06:48.09 | mut | drew |
| 06:48.11 | *** join/#asterisk morrece (n=moreece@196.46.142.23) |
| 06:48.11 | zamsler | pm me your sip.conf. |
| 06:48.16 | mut | use t |
| 06:48.18 | mut | in the dial |
| 06:48.28 | morrece | morning all |
| 06:48.38 | morrece | alrighty now where was I |
| 06:48.40 | morrece | ah yes |
| 06:49.05 | mut | ..drew? |
| 06:49.06 | morrece | my asterisk server isint providing me with port 5060 for SIP connections |
| 06:49.08 | morrece | ??? |
| 06:49.12 | zamsler | lol |
| 06:49.14 | Drew | jep im just trying that |
| 06:49.17 | zamsler | morrece, what OS? |
| 06:49.26 | morrece | suse8.2 |
| 06:49.43 | zamsler | what ver of *? |
| 06:49.54 | mut | um |
| 06:49.55 | morrece | erm 1.2.1 I think, just checking quick |
| 06:49.59 | zamsler | ok. |
| 06:50.11 | zamsler | and you have what port defined in sip.conf? |
| 06:50.12 | mut | .. |
| 06:50.20 | mut | grep port /etc/asterisk/sip.conf |
| 06:50.41 | morrece | bindport 5060 |
| 06:50.43 | zamsler | mut, you have to let them figure that out on their own.. |
| 06:50.52 | zamsler | how else will people ever get a clue.. |
| 06:50.53 | mut | zamsler: no ya don't |
| 06:50.54 | zamsler | ? |
| 06:50.57 | zamsler | lol |
| 06:51.04 | morrece | bindaddress=10.0.18.200 <-- local IP on network |
| 06:51.04 | mut | if it doesn't exist it won't show up |
| 06:51.09 | mut | if it does then it will |
| 06:51.16 | mut | either way they'll have to open it |
| 06:51.16 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 06:51.29 | morrece | ??? erm as u speaking to me? |
| 06:51.30 | zamsler | hmm |
| 06:51.36 | zamsler | no |
| 06:51.48 | zamsler | we are blabbering amongst ourselves. |
| 06:51.52 | mut | :P |
| 06:51.56 | mut | i am so teh screwed |
| 06:52.02 | mut | i can't find this g729 reg key |
| 06:52.11 | mut | and i'm putting this server online in 1.5 hrs |
| 06:52.18 | zamsler | you have problems til tomorrow. |
| 06:52.29 | Drew | mut - the t option doesnt work - it still tries to native bridge the call |
| 06:52.39 | zamsler | lol @option t |
| 06:52.42 | mut | drew: um |
| 06:52.51 | mut | are you sure asterisk isnt staying in the call? |
| 06:52.56 | mut | or don't you want it to? |
| 06:53.09 | zamsler | he wants it in the moddle |
| 06:53.22 | mut | well it's probably doing it |
| 06:53.25 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 06:53.29 | zamsler | lol |
| 06:53.40 | mut | drew: make the call and when it's going type 'show channels' |
| 06:53.48 | mut | see if anything shows |
| 06:54.11 | mut | cause it always says attempting native bridge.. |
| 06:54.50 | mut | zamsler: i use t in my dials so asterisk stays in the rtp stream and keeps audio open |
| 06:55.20 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 06:55.23 | zamsler | uhuh. |
| 06:55.58 | alephcom_ | My poor computer. I'm rating 3 months of calls. :-) It's good to see the load go up alittle on this thing. :-) |
| 06:56.23 | mut | 3 months? |
| 06:56.25 | mut | why not.. nightly? |
| 06:56.42 | Drew | right i tried the show channels - i made a call, as usual i geht about half a second of audio then the "attempting native bridging" messige appears on the console and i loose the audio - "show channels" shows 2 active channels and 1 active call |
| 06:56.54 | alephcom_ | Because that would make way to much sense. They should actually be done in realtime. |
| 06:57.05 | zamsler | uhhh huh |
| 06:57.15 | mut | drew: so asterisk is staying in the middle |
| 06:57.21 | alephcom_ | Well, I use myself for a crash dummy and I did a whole bunch of work on the rating engine so.... |
| 06:57.28 | zamsler | mut, kinda sounds like an rtp issue..? |
| 06:57.29 | mut | using t in the dial string might help your adio problem tho |
| 06:57.32 | mut | yea |
| 06:57.43 | zamsler | Drew, are these devices natted? |
| 06:57.48 | mut | drew: etiher of em natteD? |
| 06:57.50 | Drew | yes |
| 06:57.53 | mut | er yea what he said |
| 06:57.54 | zamsler | well. |
| 06:58.09 | zamsler | and in sip.conf you have nat=always ? |
| 06:58.15 | Drew | all devices are in the same network/subnet |
| 06:58.19 | reza | what does a tdm400 card look like in the dmesg output? |
| 06:58.21 | Drew | nat=yes |
| 06:58.28 | zamsler | drew.. |
| 06:58.37 | zamsler | if they are on the same networlk |
| 06:58.39 | morrece | erm ok I'm back ... so any ideas why my asterisk box isnt binding port 5060 |
| 06:58.41 | morrece | ? |
| 06:58.42 | zamsler | THEY ARE NOT NATTED... |
| 06:58.52 | *** join/#asterisk zu (n=raz@9-pool1.ras14.floca.alerondial.net) |
| 06:58.53 | zamsler | nat=no |
| 06:58.57 | mut | hell |
| 06:59.00 | mut | use nat=yes anyway |
| 06:59.14 | Drew | no - asterisk and ONE peer are on the same network |
| 06:59.24 | zamsler | I have had small issues with that mut |
| 06:59.25 | tainted- | anyone know what could cause audio not to work in asterisk? |
| 06:59.33 | zu | hy all |
| 06:59.33 | mut | small issues? |
| 06:59.38 | Drew | the other peer is a SIP account which goes through a NAT |
| 06:59.38 | tainted- | i see the audio packet going into the box |
| 06:59.40 | zamsler | yeah.. |
| 06:59.42 | tainted- | but i don't hear audio |
| 06:59.56 | zamsler | Drew, do a rtp debug |
| 07:00.06 | *** join/#asterisk NghtLght (n=ircproxy@202.179.3.171) |
| 07:00.19 | mut | bbfew |
| 07:00.22 | mut | gotta prep some shiz |
| 07:00.25 | zu | tainted-: thats usually a rtp issue |
| 07:00.27 | zu | rtp debug ip |
| 07:00.45 | Drew | what am i looking for in the debug output? |
| 07:01.03 | morrece | zamsler: what other .conf files might require configuration other than sip.conf to enable it? |
| 07:01.05 | zamsler | do you see anything after the call? |
| 07:01.11 | tainted- | zu strange.. i haven't changed anything |
| 07:01.18 | zamsler | morrece, nothing? |
| 07:01.27 | tainted- | i did install mpg123 |
| 07:01.32 | zu | what did the rtp debug do |
| 07:01.40 | zamsler | how do u know 5060 does not work? |
| 07:01.50 | mut | digium at work this late? |
| 07:02.06 | Drew | yes - rtp packets are there up to the point where the call is answerd |
| 07:02.07 | morrece | becaus when I do a port scan on the box there is no port 5060 and when I use my Xlite softphone to try make a call I get request timeout |
| 07:02.14 | mut | nope |
| 07:02.16 | zamsler | lol |
| 07:02.23 | Drew | but after answering the call no more RTP |
| 07:02.49 | zamsler | morrece, it does not show up on a portscan |
| 07:02.55 | tainted- | where does rtp debug go |
| 07:02.57 | tainted- | CLI? |
| 07:03.00 | tainted- | or a log file |
| 07:03.00 | zamsler | uhuh |
| 07:03.00 | zu | yup |
| 07:03.06 | tainted- | i don't see anything |
| 07:03.06 | zu | CLI |
| 07:03.12 | morrece | surely it should ... I mean its a UDP port |
| 07:03.13 | tainted- | rtp debug ip <ip of asterisk> |
| 07:03.23 | zu | rtp debug ip device |
| 07:03.24 | zamsler | it doesn't on my servers... |
| 07:03.41 | zamsler | and I am not doing anything special |
| 07:04.01 | zamsler | so morrece do a sip debug and open xlite . |
| 07:04.06 | zamsler | do you see anything? |
| 07:04.25 | tainted- | device meaning ip of client? |
| 07:04.25 | zamsler | morrece, did you open the port on the windows firewall? |
| 07:05.01 | zamsler | wow.. just think if people would pay for this kida support. |
| 07:05.12 | osirus | they do |
| 07:05.19 | zamsler | hmmm |
| 07:05.39 | zamsler | his ^^ |
| 07:05.50 | wasim | "MGCPGW/C1/04/18-ae4b","Zap/18-1","Dial","Zap/18","2006-01-25 02:13:50" |
| 07:05.51 | tainted- | this is weird |
| 07:05.56 | tainted- | just stop working all the sudden |
| 07:06.17 | *** join/#asterisk mgoh (n=goh@60.48.251.240) |
| 07:06.20 | reza | zamsler - wrong module :P |
| 07:06.36 | reza | wasted an hour because you said it was wct4xxp; it's the wctdm!! |
| 07:06.38 | reza | grr |
| 07:06.39 | rob0 | Or the kernel is tainted- :) |
| 07:06.53 | tainted- | how do i completely remove mpg123? |
| 07:06.59 | tainted- | that's the only thing that has changed |
| 07:07.02 | justinu | what's chan_mgcpgw? |
| 07:07.08 | zu | tainted use native |
| 07:07.18 | wasim | justinu: its user agent mode for * |
| 07:07.21 | mut | anyone around from digium can get me my g729a reg key? |
| 07:07.25 | justinu | oooh, nice |
| 07:07.26 | tainted- | zu native? |
| 07:07.27 | zu | then convert the mp3 to the prefered codec |
| 07:07.28 | mut | or know how to retrieve it from my current install? |
| 07:07.43 | wasim | justinu: yeah, long time coming ... finally got it done |
| 07:07.44 | tainted- | this was all working fine last night |
| 07:07.55 | wasim | justinu: anthm to the rescue :) |
| 07:08.18 | morrece | *CLI> rtp debug 10.0.18.200:5060] |
| 07:08.24 | morrece | *CLI> rtp debug 10.0.18.200:5060 |
| 07:08.32 | justinu | wasim: does that guy ever stop? ;) |
| 07:08.33 | zu | tainted see: /etc/asterisk/musiconhold.conf |
| 07:08.45 | wasim | justinu: for all our sakes, i pray not |
| 07:08.52 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 07:08.54 | justinu | wasim: agreed :) |
| 07:08.55 | zu | no 5060 needed |
| 07:09.02 | mut | grr |
| 07:09.07 | mut | no one from digium around |
| 07:09.09 | zu | 5060 is sip messages |
| 07:09.23 | zu | rtfm rfc3261 |
| 07:09.25 | morrece | no it tells me to put it it |
| 07:09.30 | zu | ~rfc3261 |
| 07:09.32 | mut | they need like a form on their website to email keys |
| 07:09.33 | morrece | Usage: rtp debug [ip host[:port]] |
| 07:09.33 | morrece | <PROTECTED> |
| 07:09.43 | zu | just do the host |
| 07:09.47 | wasim | morrece: 5060 is not an rtp port |
| 07:09.55 | morrece | I have and I get the usage |
| 07:10.03 | morrece | *usuage message* |
| 07:10.18 | morrece | rtp debug 10.0.18.200 -> I get the usuage message |
| 07:10.26 | zu | rfc3261 sip http://www.ietf.org/rfc/rfc3261.txt |
| 07:10.29 | tainted- | morrece what error are u having |
| 07:10.47 | morrece | oops |
| 07:10.52 | tainted- | zu host w/o port specified? |
| 07:10.57 | justinu | jbot, rfc3261 is a document explaining sip |
| 07:10.58 | jbot | okay, justinu |
| 07:10.59 | tainted- | i don't get any rtp debug |
| 07:11.00 | morrece | didnt put ip in like rtp debug ip 10.0.18.200 |
| 07:11.04 | morrece | ok its enabled |
| 07:11.05 | zu | yup |
| 07:11.07 | morrece | going to try xlite |
| 07:11.08 | *** join/#asterisk remmos (n=chatzill@adsl-1180.isp.net.au) |
| 07:11.16 | remmos | where is the app StipMSD ? |
| 07:11.36 | ManxPower | remmo, removed. See README.variables on how to strip digits |
| 07:11.39 | wasim | remmos: she's taking the day off today, will be back the club tomorrow |
| 07:12.02 | remmos | i know how to strip digits. i have an agi app and need stipmsd |
| 07:12.23 | wasim | remmos: use the power of the programming language of your agi |
| 07:12.47 | remmos | that was something i was trying to avoid. do you know when it was depreciated? |
| 07:12.54 | morrece | ok nothing on the console: when using Xlite to connect |
| 07:13.05 | morrece | like its not even making tha connection |
| 07:13.16 | morrece | doesnt it only move over to RTP when connection is established? |
| 07:13.18 | tainted- | zu RTP Debugging Enabled for IP: 71.134.157.119 |
| 07:13.25 | tainted- | but i don't see any packets at all |
| 07:13.26 | zu | then initate a call |
| 07:13.30 | tainted- | during a call |
| 07:13.49 | morrece | ok, gotta run quick ... bosss needs me |
| 07:13.54 | morrece | will be back to further discuss |
| 07:13.57 | morrece | thanks all |
| 07:14.35 | zu | jbot, Substrings ${foo:offset:length} returns a substring of the string foo, beginning at offset offset and returning the next length characters. |
| 07:16.03 | NewSole | ok is there a reason why we are getting no audio on sip or IAX2 now |
| 07:16.06 | zu | tainted then try a tcpdump -i ethernet interface |
| 07:16.27 | tainted- | i see the udp packets |
| 07:16.30 | zu | or tcpdump src host (IP ADDR) -i (ethernet device) |
| 07:16.36 | tainted- | NewSole are u experiencing the same thing? |
| 07:16.57 | tainted- | no audio on incoming or outgoing SIP or IAX2 |
| 07:17.15 | zu | iax2 2 |
| 07:17.26 | NewSole | yup |
| 07:17.37 | tainted- | NewSole did this just happen today? |
| 07:17.44 | NewSole | yup |
| 07:17.45 | tainted- | it just started happening today for me |
| 07:17.56 | tainted- | even internal calls |
| 07:18.01 | NewSole | yup |
| 07:18.01 | tainted- | voip end to end |
| 07:18.10 | tainted- | shit |
| 07:18.15 | tainted- | sounds like an asterisk bug then |
| 07:18.20 | zu | weird, was there like a auto firmware update on the device? |
| 07:18.29 | tainted- | no |
| 07:18.37 | tainted- | i'm tryign various devices/softphones |
| 07:18.47 | NewSole | i debuged iax2 and was not sending any voice frames |
| 07:18.51 | tainted- | polycom 301s, gs 488s |
| 07:19.07 | zu | do they ring? |
| 07:19.11 | tainted- | yes they ring |
| 07:19.17 | tainted- | and u can pick up calls |
| 07:19.20 | tainted- | but no audio at all |
| 07:19.28 | zu | can you see those 5060 udp packets? |
| 07:19.32 | tainted- | yes |
| 07:19.45 | zu | have ya rebooted lately? |
| 07:19.48 | tainted- | yes |
| 07:19.51 | tainted- | twice |
| 07:19.53 | NewSole | same |
| 07:20.01 | zu | what distro? |
| 07:20.08 | tainted- | NewSole did u add mpg123 recently? |
| 07:20.12 | tainted- | fedora core 4 here |
| 07:20.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:20.29 | zu | do you have a yum autoupdate in your crontabs? |
| 07:20.38 | tainted- | no |
| 07:20.45 | *** join/#asterisk EriSan (n=erisan@151.8.109.86) |
| 07:20.45 | tainted- | i do all dist-upgrades manually |
| 07:20.58 | NewSole | nope still running core3 |
| 07:21.09 | zu | what version of * |
| 07:21.17 | NewSole | 1.2.2 |
| 07:21.20 | tainted- | 1.2.2 |
| 07:21.32 | tainted- | it has affected all production servers |
| 07:21.35 | zu | svn co the cvs |
| 07:21.36 | zu | now |
| 07:21.43 | justinu | anyone know if the TE210P card does hardware DTMF recognition? |
| 07:21.52 | ManxPower | tainted-, what's happening? |
| 07:22.03 | tainted- | no audio on any voip channels |
| 07:22.09 | ManxPower | justinu, I think only if it has the echocan module. |
| 07:22.13 | tainted- | i see UDP, don't hear any audio |
| 07:22.17 | tainted- | no moh either |
| 07:22.22 | tainted- | two separate boxes |
| 07:22.23 | NewSole | nothin |
| 07:22.24 | wasim | yeah, with the DSP innit |
| 07:22.25 | justinu | ManxPower: any idea how I can determine that? |
| 07:22.36 | tainted- | different softclients and IP phones |
| 07:22.44 | tainted- | separate networks |
| 07:22.49 | ManxPower | justinu, Well since they are like $500 extra, I think you would know. |
| 07:22.51 | reza | ok |
| 07:23.01 | wasim | :) |
| 07:23.04 | justinu | yeah, i don't really... i didn't buy this card |
| 07:23.08 | reza | i'm getting close to getting this thing to work.. |
| 07:23.15 | justinu | and i'm remote to the box |
| 07:23.21 | ManxPower | justinu, It's a daughter card, I think. |
| 07:23.22 | wasim | dmesg |
| 07:23.22 | reza | i've got the tdm400, and using NuFone I can call a number |
| 07:23.24 | ManxPower | justinu, call digium |
| 07:23.26 | tainted- | NewSole i thought i was the only one going crazy |
| 07:23.31 | reza | but when someone picks up the phone, i can't hear anything either way |
| 07:23.38 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-123.claranet.co.uk) |
| 07:23.42 | mut | no ones home at this early hour |
| 07:23.52 | mut | cause i need my g729codec registration # |
| 07:23.56 | tainted- | reza did this happen recently? |
| 07:23.57 | mut | so i can put the codec on a new machine |
| 07:24.20 | reza | just now |
| 07:24.27 | tainted- | same |
| 07:24.36 | tainted- | both NewSole and i just had the same issue crop up |
| 07:24.40 | Drew | reza - tainted - i have the same problem - since i rebooted the *-box today |
| 07:24.45 | ManxPower | So is this a newfone issue? |
| 07:24.47 | reza | what? problem with nufone? |
| 07:24.50 | tainted- | no |
| 07:24.54 | tainted- | i don't even use nufone |
| 07:25.07 | morrece | ok back yet again |
| 07:25.14 | tainted- | no audio on local calls |
| 07:25.16 | tainted- | either |
| 07:25.19 | Drew | i first thought it was a change in a config but my old working ones dont work anymore either |
| 07:25.26 | tainted- | morrece i think we're all experiencing the same issue |
| 07:25.34 | ManxPower | I've not had any reports from the users of the 6 asterisk systems I manage |
| 07:25.39 | tainted- | Drew i changed nothing and it abruptly stopped working |
| 07:25.45 | tainted- | ManxPower 1.2.2? |
| 07:25.52 | ManxPower | tainted-, HAHAHAH!!!!!! |
| 07:26.04 | tainted- | what |
| 07:26.07 | Drew | tainted - what is your rtp debug output? |
| 07:26.09 | NewSole | this happen between 2-8 pm today |
| 07:26.12 | ManxPower | 1.2.2. SURE, I upgrade all my customers as soon as a new release comes out! I don't even test it first! |
| 07:26.16 | Drew | what timezone? |
| 07:26.20 | NewSole | EST |
| 07:26.22 | mut | ManxPower: know how to grab the reg # for my 729 codec from a machine with it installed |
| 07:26.26 | mut | or is it lost til digium can send it to me? |
| 07:26.33 | tainted- | ManxPower what version are u running then |
| 07:26.49 | reza | tainted - so you are having the same problem? |
| 07:26.51 | tainted- | ManxPower i recluctantly upgraded after being refused support |
| 07:26.53 | ManxPower | tainted-, 1.2.0, 1.2.RC2, 1.0.9. |
| 07:26.57 | tainted- | reza yea |
| 07:27.01 | reza | is it a config problem or a sip/aix provide rproblem? |
| 07:27.03 | tainted- | i was running 1.0.7 |
| 07:27.09 | morrece | another question, with regards to my SIP phone settings, my SIP proxy is my asterisk box IP and my realm is simply asterisk |
| 07:27.15 | tainted- | i think it's a 1.2.2 problem |
| 07:27.20 | reza | grr |
| 07:27.42 | tainted- | ManxPower is 1.2.0 stable? |
| 07:27.46 | reza | latest cvs? |
| 07:27.57 | tainted- | i think i'm about to downgrade my boxes RIGHT NOW |
| 07:28.03 | *** join/#asterisk hnr (n=hnr@host116-43.pool8174.interbusiness.it) |
| 07:28.04 | ManxPower | tainted-, Seems to be stable enough. |
| 07:28.09 | reza | is there an easy way to downgrade w/ cvs? |
| 07:28.25 | ManxPower | And my users are whiney assholes, so if it wasn't, I'd know about it. |
| 07:28.26 | tainted- | just remove files in modules folder right? |
| 07:28.30 | morrece | how do I quickly check my version, I think I am 1.2.1 |
| 07:28.39 | ManxPower | "show version" |
| 07:28.40 | tainted- | and then build 1.2.0? |
| 07:28.49 | hnr | hi, how to know how many second my agi c program have used ? (sorry for bad english) ? |
| 07:28.49 | morrece | ah simply enough |
| 07:28.54 | morrece | ta |
| 07:29.34 | morrece | hmmm so tainted u reckon we are having the same issue? |
| 07:29.47 | Drew | ok im out for coffee and jogging around the block for some fresh air - my brain cant stand anymore *-issues for the moment |
| 07:29.53 | reza | i'm running Asterisk SVN-trunk-r7230 |
| 07:30.05 | budmang | http://cgi.ebay.com/Zyxel-WiFi-Wireless-VOIP-SIP-IP-Phone-802-11b-Asterisk_W0QQitemZ5858267050QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem#ebayphotohosting |
| 07:30.17 | tainted- | reza, morrece, NewSole, Drew what codec do u guys use |
| 07:30.34 | reza | it defaulted to gsm |
| 07:30.43 | reza | i'm just trying to get the setup working |
| 07:30.58 | Drew | g177a |
| 07:31.18 | tainted- | damn i'm stupid |
| 07:31.27 | tainted- | should've stuck with 1.0.7 |
| 07:31.35 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 07:31.43 | morrece | my show codecs reveals the following |
| 07:31.45 | morrece | http://pastebin.com/521922 |
| 07:31.53 | morrece | its all defaulted |
| 07:32.40 | morrece | although why would the codecs stop it from connecting all together, I understand once connection is established that codes could play a part however I'm not even establishing that |
| 07:33.03 | morrece | with regards to the sip phone, doesnt it need to login to the sip proxy which is asterisk??? |
| 07:33.09 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:33.17 | tainted- | morrece are u running 1.2.2 |
| 07:33.18 | mut | oh hell yea |
| 07:33.22 | morrece | 1.2.1 |
| 07:33.24 | mut | the registration key is the filename! |
| 07:33.29 | mgoh | who can give me some info. can asterisk work with dual processor. any configure that I need to make or just like normal install. |
| 07:33.48 | mut | /var/lib/asterisk/licenses/*.lic |
| 07:33.54 | tainted- | mut u sure about that |
| 07:34.10 | tainted- | mut i think it's tied to your box through some kind of serial |
| 07:34.17 | mut | yes |
| 07:34.24 | mut | but the key to register it again is the filename |
| 07:34.28 | mut | cause i'm moving it to a new box |
| 07:34.31 | tainted- | really |
| 07:34.35 | tainted- | how do u re-register |
| 07:34.44 | mut | just run ./register again |
| 07:34.51 | mut | ya get to do it only once after initial register |
| 07:34.55 | mut | then ya gotta call em |
| 07:35.04 | tainted- | hmm |
| 07:35.13 | tainted- | they told me that they don't let me transfer licenses |
| 07:35.17 | NewSole | have we figured out why no audio yet |
| 07:35.23 | reza | licenses of what? codec? |
| 07:35.26 | mut | ya |
| 07:35.28 | mut | 729a |
| 07:35.34 | reza | worth it? |
| 07:35.37 | tainted- | NewSole i'm rolling back to 1.2.0 to find out |
| 07:35.52 | reza | tainted - how are you rolling back? where are you geting the 1.2.0 source? |
| 07:35.59 | mut | see, just re-regged it |
| 07:36.00 | mut | and it says |
| 07:36.00 | mut | REMINDER: You will NOT be able to re-register this key with a different |
| 07:36.01 | mut | Host-ID without contacting Digium first! |
| 07:36.06 | mut | O_o |
| 07:36.20 | tainted- | http://ftp.digium.com/pub/asterisk/old-releases/ |
| 07:36.33 | tainted- | mut must be a new policy |
| 07:36.39 | mut | uh |
| 07:36.42 | mut | i got my key 2 yr ago |
| 07:36.44 | reza | just found it right before you pasted ;) |
| 07:36.46 | mut | and it was in effect then |
| 07:37.13 | reza | is it just asterisk that you're recompiling or zaptel/libpri as well? |
| 07:37.23 | mut | it's in the frickin readme |
| 07:37.29 | tainted- | HEY ALL |
| 07:37.38 | tainted- | ROLLING BACK TO 1.2.0 FIXES AUDIO PROBLEMS |
| 07:37.46 | mut | i have no audio problems |
| 07:37.49 | mut | i have svn of today.. |
| 07:38.08 | reza | .. slow download :( |
| 07:38.17 | tainted- | well if you were experiencing SIP/IAX2 audio issues, try 1.2.0.. worked for me |
| 07:38.21 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 07:38.25 | mut | oo |
| 07:38.33 | mut | didn't test w/ iax |
| 07:38.52 | mut | just sip -> zap and the reverse and sip -> sip |
| 07:38.53 | *** join/#asterisk rajiv_ (n=irc@gentoo/developer/rajiv) |
| 07:39.33 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:40.09 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 07:40.29 | NewSole | Digium put a time bomb in code..... |
| 07:40.31 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 07:40.38 | Drew | lol |
| 07:40.54 | NewSole | tainted- what time did you die |
| 07:41.00 | Drew | what about setting the system time back ;-) |
| 07:41.21 | reza | is it possible (or does this exist) - some way of setitng up phone prompting such taht you can type in the caller id number you want displayed, then on the next call, it will use that number? |
| 07:41.32 | tainted- | NewSole i'd say around 9PM PST |
| 07:41.45 | NewSole | yup about same time mine did |
| 07:41.50 | mut | er wtf |
| 07:41.58 | mut | mine isn't workin either |
| 07:41.59 | Drew | problems since reboot - my uptime is aprox 4h |
| 07:42.16 | mut | it was this morning when i tested it |
| 07:42.29 | mut | but that was like 10:30 est |
| 07:42.32 | reza | glad i asked first; would have sucked to run around trying to fix a non-existant config problem |
| 07:42.45 | reza | i'm glad i put a faster processor in this box; compling is going faster |
| 07:42.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 07:42.58 | NewSole | it happen to us at between 2-8pm EST |
| 07:43.44 | NewSole | mut new svn dont help i tried that |
| 07:43.58 | NewSole | BEFORE i cam on IRc |
| 07:44.12 | reza | so if this works i will be very impressed and happy |
| 07:44.29 | tainted- | hmm |
| 07:45.00 | mut | no way |
| 07:45.02 | mut | it's time based |
| 07:45.06 | mut | i changed my system date to 2005 |
| 07:45.08 | mut | and audio works |
| 07:45.09 | reza | what's chan_phone.so and why does it crash asterisk |
| 07:45.18 | tainted- | mut ARE YOU SERIOUS |
| 07:45.23 | mut | yes |
| 07:45.32 | mut | lemme change it back and test again |
| 07:45.33 | mut | sed |
| 07:45.34 | mut | sec |
| 07:45.34 | tainted- | that is a timebomb |
| 07:46.06 | NewSole | must be Digiums way to Get $$ for support |
| 07:46.09 | mut | yep |
| 07:46.11 | mut | time based |
| 07:46.15 | mut | changed back to 06 and no work |
| 07:46.32 | tainted- | hmm |
| 07:46.34 | NewSole | I reverted to older build and now works |
| 07:46.45 | Drew | ROFL it works in 2005 |
| 07:47.17 | reza | hrmph |
| 07:47.32 | Drew | *turning off ntpd* |
| 07:48.26 | mut | i just |
| 07:48.29 | mut | ran date |
| 07:48.32 | mut | then copied to |
| 07:48.42 | mut | date -s 'pasted & changed to 05' |
| 07:48.50 | reza | 1.2.0 dies when it gets to chan_zo.so |
| 07:48.51 | reza | thoughts? |
| 07:49.09 | tainted- | lol |
| 07:49.09 | Drew | update to 1.2.2 ;-) |
| 07:49.15 | tainted- | here's the kicker |
| 07:49.19 | tainted- | anyone here use teliax? |
| 07:49.33 | tainted- | they are on 1.2.2 and it has DEFINITELY affected their systems |
| 07:50.01 | Drew | my sip provider seams to still be funtioning... |
| 07:50.01 | tainted- | reza clear out your /usr/lib/asterisk/modules before compiling |
| 07:50.10 | reza | doi |
| 07:50.14 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:50.31 | reza | module got unloaded; reinstalling 1.2.0 and testing |
| 07:50.41 | reza | why is it recompling; grr.. |
| 07:50.52 | Drew | is a admin here to change the channel topic?... probably there will be a lot of ppl affected by this |
| 07:53.10 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com) |
| 07:53.38 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:54.57 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:56.22 | reza | tainted - 1.2.0 still doesn't help |
| 07:56.24 | reza | grr |
| 07:56.34 | tainted- | local calls? |
| 07:56.39 | tainted- | or calls to provider |
| 07:56.48 | reza | call to NuFone |
| 07:57.01 | reza | land line -> TDM400 -> NuFone -> my cell phone |
| 07:57.04 | tainted- | well if nufone is on 1.2.2 he is equally fucked |
| 07:57.31 | tainted- | my local calls work now |
| 07:57.37 | mut | alright |
| 07:57.37 | tainted- | but teliax is down |
| 07:57.38 | mut | heres what i got |
| 07:57.42 | reza | hmm |
| 07:57.50 | *** join/#asterisk graabein (n=gunnar@bkkb-gw.bitcon.no) |
| 07:57.51 | mut | foghorn:~# date -s 'Wed Jan 25 00:18:36 EST 2006' |
| 07:57.53 | mut | audio workjs fine |
| 07:57.57 | reza | is there some other provider i can use for free to test? |
| 07:57.58 | mut | foghorn:~# date -s 'Wed Jan 25 00:19:00 EST 2006' |
| 07:58.04 | mut | audio does not work after this point |
| 07:58.15 | mut | and time before Wed Jan 25 00:18:36 EST 2006 it works though |
| 07:58.30 | tainted- | mut that's crazy |
| 07:58.30 | Drew | http://bugs.digium.com/view.php?id=6349 <= bugreport to the SIP/Date/Audio problem |
| 07:58.48 | mut | i don't even have to restart the server either |
| 07:58.52 | mut | i just set date and make a call |
| 07:59.28 | tainted- | hmm |
| 08:00.04 | tainted- | maybe digium's way of estimating asterisk adoption rate? :D |
| 08:00.20 | mut | woah! |
| 08:00.29 | reza | anyong using gsm audio? |
| 08:01.03 | Drew | maybe some evil conspiricy ;-) |
| 08:01.26 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 08:01.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:02.29 | mut | crazy! |
| 08:02.36 | mut | after |
| 08:02.36 | mut | date -s 'Wed Jan 25 00:18:45 EST 2006' |
| 08:02.38 | mut | er |
| 08:02.43 | mut | Wed Jan 25 00:18:52 EST 2006 |
| 08:02.44 | mut | audio dies |
| 08:02.47 | mut | while i'm on the phone even |
| 08:02.54 | Drew | rofl |
| 08:02.54 | reza | tainted- so what provider do you use and have it work? |
| 08:03.01 | mut | anytime before that it works fine |
| 08:03.06 | mut | so.. wtf? |
| 08:03.08 | mut | heh |
| 08:03.10 | Drew | is it timezone dependent? |
| 08:03.15 | mut | lemme see |
| 08:03.31 | Drew | probably the unix epoch date counts and not your local time |
| 08:03.57 | mut | well what time is that unix time? |
| 08:04.22 | reza | ? |
| 08:04.34 | reza | unix time = seconds since jan 1 1970, i think |
| 08:04.41 | reza | i think it's gmt |
| 08:04.48 | mut | 1138166332 |
| 08:04.52 | Drew | EST is UTC+5 ?? |
| 08:04.59 | mut | -5 |
| 08:05.18 | Drew | thats what i meant ;-) |
| 08:05.18 | reza | i think +, no? |
| 08:05.23 | reza | guess not |
| 08:05.38 | reza | so, for those that have managed to get it to work - |
| 08:05.49 | reza | what works? what are you doing? |
| 08:05.53 | reza | calling through a provider? |
| 08:06.02 | mut | i just have a server setup |
| 08:06.05 | mut | and a sip phone |
| 08:06.10 | mut | dial in/out a pri |
| 08:06.19 | mut | server is supposed to go online in 24 min |
| 08:06.21 | reza | can you dial out and get it owrking? |
| 08:06.38 | reza | like call a land line from your setup |
| 08:06.53 | mut | thats what i was doing to test it |
| 08:07.03 | reza | what provider do you us? |
| 08:07.04 | reza | use? |
| 08:07.07 | mut | none |
| 08:07.27 | reza | you have an fxo card? |
| 08:07.40 | Drew | isnt it a bit late for Y2K bugs? |
| 08:07.46 | *** join/#asterisk bmg505 (n=leon@dsl-146-35-67.telkomadsl.co.za) |
| 08:08.44 | mut | no |
| 08:09.04 | mut | te405p |
| 08:09.07 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:09.35 | mut | this is just nuckin futz |
| 08:10.07 | reza | [root@dhcp90 zaptel]# perl -e 'print time,"\n"' |
| 08:10.07 | reza | 1137734923 |
| 08:10.08 | tainted- | yea |
| 08:10.39 | RoyK | hm, Drew?, anyone? is asterisk rfc2550 compliant? |
| 08:11.42 | Drew | lol |
| 08:11.56 | Drew | RoyK look at the date of that RFC ;-) |
| 08:12.16 | RoyK | i know :) |
| 08:12.40 | Drew | well its a Y2K6 problem not a Y10K problem ;-) |
| 08:13.08 | reza | when you experience the problem - you just get no audio at all, eh? |
| 08:13.10 | RoyK | where? |
| 08:14.05 | Drew | yes reza - a soon as the time goes past unix timestamp 1138166332 the RTP stream gets killed |
| 08:14.15 | Drew | or thats what appears to be happening |
| 08:14.30 | RoyK | when did this happen, Drew? |
| 08:14.37 | mut | there |
| 08:14.38 | Drew | 1138166332 |
| 08:14.39 | mut | aded my notes to the bug |
| 08:14.57 | Drew | *thumbs up* @ mut |
| 08:15.07 | mut | now |
| 08:15.10 | RoyK | Drew: related to zaptel? |
| 08:15.14 | mut | i gotta install a older version |
| 08:15.20 | RoyK | Drew: happened quite recently? |
| 08:15.22 | mut | cause this is sposed to be online in 15 min |
| 08:15.34 | mut | RoyK: it happened at exactly Wed Jan 25 00:18:52 EST 2006 |
| 08:15.46 | RoyK | zaptel 1.2 related |
| 08:15.47 | RoyK | ? |
| 08:16.04 | Drew | no idea |
| 08:16.07 | mut | may be? my test calls were using a zap chan |
| 08:16.20 | mut | but it was svn checkout of today |
| 08:16.22 | Drew | but i am using only SIP channels |
| 08:16.29 | Drew | it i get the same bug |
| 08:16.31 | reza | ok, testing with the demo connection |
| 08:16.36 | RoyK | i just had problems, tonight, that suddenly we got no sound from Zaptel to IAX2 |
| 08:16.41 | *** join/#asterisk budmang (i=budman@12-216-205-236.client.mchsi.com) |
| 08:16.51 | Drew | when tonight? |
| 08:16.54 | mut | i think it's an all over channel affecting 'bug' |
| 08:16.59 | *** join/#asterisk sergey (n=sergey@gw3-a16.iks.ru) |
| 08:17.05 | Drew | about 5h ago? |
| 08:17.53 | *** join/#asterisk pardove (n=pardove@217.219.250.24) |
| 08:18.17 | *** join/#asterisk SOLARMASTER (n=SOLARMAS@Quebec-HSE-ppp215813.qc.sympatico.ca) |
| 08:18.20 | pardove | no audio on * since 01/25/2006 |
| 08:18.30 | Drew | http://bugs.digium.com/view.php?id=6349 |
| 08:18.43 | Drew | set the date back ;-) |
| 08:19.01 | *** join/#asterisk budman2 (i=budman@12-216-205-236.client.mchsi.com) |
| 08:19.04 | RoyK | Drew: timezone what? |
| 08:19.12 | tainted- | fuck |
| 08:19.36 | mut | RoyK: test it... |
| 08:19.36 | mut | just |
| 08:19.45 | mut | change the year to 2005 |
| 08:19.48 | mut | and make a test call |
| 08:19.58 | reza | hmm, audio works when i aix to the aix server at digium |
| 08:20.06 | reza | it might be a problem with a config file.. |
| 08:20.07 | Drew | 2006 JAN 25 - 05:18:52 UTC |
| 08:20.34 | Drew | i dont know what timezone you r in roy - so i can tell you UTC |
| 08:21.06 | mut | drew: can y test it? |
| 08:21.09 | mut | you* |
| 08:21.16 | mut | see if it;s the same time for you.. |
| 08:21.21 | Drew | hang on |
| 08:21.24 | mut | maybe it's dependant on something in the system too? |
| 08:21.35 | mut | cause thats an aweful weird time.. |
| 08:21.58 | mut | you say 1.2.2 doesn't work or does work though? |
| 08:22.20 | reza | does this look sane : http://pastebin.com/521958 |
| 08:23.22 | budman2 | can someone help me connnect with my voip behind a router |
| 08:23.26 | budman2 | softphone |
| 08:23.50 | RoyK | mut: that box is down, and will stay down until i've downgraded it to 1.0 (that is, swapped it to an 1.0 box) |
| 08:23.56 | pardove | is there any bugmarshal in? |
| 08:24.04 | RoyK | this is a 10 PRI production system, so i don't just 'test' stuff |
| 08:24.06 | SOLARMASTER | If you are bored of IRC, take time to join my game at : http://sre.servegame.com, I need some beta-testers who like strategics webgames (Its a port/clone of SRE[SOLAR REALMS ELITE]), come and play |
| 08:24.16 | *** part/#asterisk SOLARMASTER (n=SOLARMAS@Quebec-HSE-ppp215813.qc.sympatico.ca) |
| 08:24.31 | RoyK | so much for nickserv requirements |
| 08:24.35 | Drew | mut - tested |
| 08:25.07 | Drew | audio gets killed exaclty the same UTC/*nix time |
| 08:25.11 | mut | k |
| 08:25.15 | *** join/#asterisk budmang (n=budman@12-216-205-236.client.mchsi.com) |
| 08:25.20 | mut | rtp debug shows nothing other than rtp stopping |
| 08:25.27 | Drew | => its not timezone dependant |
| 08:25.31 | Drew | jup |
| 08:26.05 | mut | you're testing sip -> sip correct? |
| 08:26.51 | Drew | i am testing Local softpone => sip provider |
| 08:27.12 | mut | k no zap involved then |
| 08:27.13 | Drew | softphone |
| 08:27.30 | Drew | no - zaptel dummy modules maybe for timing but i dont know |
| 08:27.35 | Drew | no zaptel channels |
| 08:29.46 | Drew | digitum issue tracker signup page is also broken |
| 08:30.42 | *** join/#asterisk _Traveller_ (n=Raiden@anzhero.ru) |
| 08:30.54 | *** part/#asterisk _Traveller_ (n=Raiden@anzhero.ru) |
| 08:34.16 | *** join/#asterisk pardove (n=pardove@217.219.250.24) |
| 08:34.20 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3781758.sympatico.ca) |
| 08:34.43 | mut | this fsckin blows |
| 08:35.59 | mut | is the issue in 1.2.2? |
| 08:36.06 | Drew | why r u checking your filesystem? its got to do with time :D |
| 08:36.06 | mut | or just latest svn |
| 08:36.27 | mut | heh |
| 08:36.41 | mut | maybe if i only load like zap/sip modules |
| 08:36.49 | mut | and ulaw codec |
| 08:36.59 | mut | and like.. basic basic modules |
| 08:37.04 | pardove | btw, it a time issue |
| 08:37.15 | RoyK | mut: i just use basic modules and only use zap/iax2 |
| 08:37.27 | RoyK | mut: so it's not related to rtp/sip |
| 08:37.34 | mut | never said it was |
| 08:39.01 | mut | wtf |
| 08:39.05 | mut | compiled 1.2.1 and i get Jan 25 00:27:04 WARNING[20774]: loader.c:142 ast_unload_resource: Firm unload failed for app_cut.so |
| 08:39.05 | mut | Jan 25 00:27:04 WARNING[20774]: loader.c:554 load_modules: Loading module app_cut.so failed! |
| 08:39.51 | pardove | remove app_cut.so |
| 08:40.04 | mut | yeh |
| 08:40.10 | pardove | from /usr/lib/asterisk/modules |
| 08:40.25 | mut | well |
| 08:40.29 | mut | audio works on 1.2.1 |
| 08:40.34 | mut | hopefully everything else does too |
| 08:40.40 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:40.54 | pardove | check your date is right |
| 08:41.04 | mut | yea it is |
| 08:41.31 | Drew | i think ill stay with 1.2.2 - i dont care if the date is wrong - its only my home PBX ;) |
| 08:41.46 | mut | everything looks good |
| 08:41.53 | *** join/#asterisk _4d4m_ (n=adam@34-14-101-159.adsl.legend.co.uk) |
| 08:41.59 | mut | bbfew |
| 08:43.26 | pardove | a diff of chan_sip version from 1.2.1 to 1.2.2 may help |
| 08:45.26 | RoyK | pardove: it's not SIP! |
| 08:45.31 | RoyK | pardove: it's prolly zap |
| 08:45.44 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:47.02 | pardove | but also sip -> sip calls doesn't work |
| 08:47.16 | RoyK | then it's not a chan_ |
| 08:47.57 | pardove | maybe an rtp issue |
| 08:48.03 | RoyK | no |
| 08:48.04 | RoyK | it's not |
| 08:48.11 | *** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar) |
| 08:48.15 | RoyK | pardove: neither iax2 nor zap uses rtp |
| 08:48.33 | RoyK | pardove: and as i said, we experience the same problem on a zap/iax2 gw |
| 08:48.41 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 08:48.47 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 08:48.48 | pardove | so channel.c |
| 08:49.01 | RoyK | sounds reasonable |
| 08:51.00 | Drew | i remember reading something about zaptel dummy modules used for timing... that could be a source |
| 08:51.03 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:51.25 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 08:53.51 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 08:54.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@82.166.242.248) |
| 08:56.46 | jpk | Hi. |
| 08:57.01 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:57.02 | jpk | Strange problem. My * does not provide ANY audio on bridged calls anymore. |
| 08:57.18 | jpk | Everything is ok up to the bridging (voiceprompts ok etc.) but after that: nothing. |
| 08:57.27 | jpk | any idea? |
| 08:57.36 | Drew | set your date back |
| 08:57.44 | Drew | to anything before today |
| 08:58.17 | jpk | excuse me? |
| 08:58.24 | Drew | you heard me right |
| 08:58.25 | Drew | http://bugs.digium.com/view.php?id=6349 |
| 08:58.39 | Drew | its a bug - audio doesnt work anymore since today |
| 08:59.30 | jpk | Now THAT is interesting. |
| 08:59.31 | jpk | *trying* |
| 08:59.42 | Drew | *nix timestamp 1106612332 |
| 09:00.05 | Drew | after the * system passes that timestamp the audio of all calls will fail |
| 09:01.04 | fdask | lol what |
| 09:01.25 | Drew | no shit - i tested that |
| 09:01.27 | mut | 2 pri's |
| 09:01.37 | mut | span=1 & span=2 on a te405p |
| 09:01.38 | [av]bani | holy fucking shit, thats whats jacking up mine too? |
| 09:01.42 | mut | for my bchan and dchan |
| 09:01.42 | mut | bchan=1-23,25-47 |
| 09:01.42 | mut | dchan=24,48 |
| 09:01.44 | mut | right? |
| 09:01.45 | *** join/#asterisk felix (n=Miranda@local.epyron.de) |
| 09:02.16 | [av]bani | god damn, i was trying to figure out why teliax and JN were no longer working |
| 09:02.19 | mut | or so i make the span |
| 09:02.22 | *** join/#asterisk elcuco (n=diego@82.166.242.248) |
| 09:02.23 | mut | then define the chans |
| 09:02.28 | oej | This with the tarballs release or svn 1.2? |
| 09:02.45 | Drew | i installed mine a few days ago off a tarball |
| 09:02.56 | oej | Ok, so both then |
| 09:03.00 | Drew | yep |
| 09:03.04 | JunK-Y | which is really bad. |
| 09:03.11 | [av]bani | so i'm assuming there will be a 1.2.3 in a few hours then |
| 09:03.14 | oej | We need to trace the frames and see what is up |
| 09:03.19 | [av]bani | this is massive badness |
| 09:03.20 | e1cuc0 | hi :) |
| 09:03.32 | Drew | this whole bug is really very bad - especially for productive systems |
| 09:03.32 | oej | Does it affect only asterisk bridging or native bridges as well? |
| 09:03.36 | oej | Like when SIP re-invite? |
| 09:03.44 | oej | Let's try to fix it then |
| 09:03.46 | Drew | i cant test that - im behind a nat |
| 09:03.48 | [av]bani | Drew: yeah, basically every * box on the entire planet is dead now? |
| 09:03.48 | RoyK | Drew: don't just 'set your date back'. fix the bug :P |
| 09:04.05 | oej | Ok, people, let's work together. |
| 09:04.24 | oej | Please test which channels this affect and whether it's only asterisk core bridging or all bridges |
| 09:04.27 | felix | hi, we are using asterisk 1.2.1, everything went fine, but today * is just not bridging any audio data |
| 09:04.34 | jpk | re. Audio is back... |
| 09:04.38 | oej | Developers: Try to figure out how a DATE can affect operation of bridges |
| 09:04.38 | jpk | That is one strange bug! |
| 09:04.39 | Drew | roy - right - but my c/c++ knowleage isnt that good and ive only been using * since 3 days |
| 09:04.54 | felix | when I call a internal moh i can here music, but besides that nothing |
| 09:05.04 | oej | moh is not bridge |
| 09:05.08 | oej | So it's bridges then |
| 09:05.11 | felix | wether on Zap (Pri) nor Sip |
| 09:05.13 | Drew | felix - moh is not bridged |
| 09:05.17 | jpk | fn~oej: It affects ALL bridged calls. The moment things get bridged, audio is dead |
| 09:05.27 | fdask | local bridges too? |
| 09:05.29 | oej | jpk: Thanks |
| 09:05.30 | jpk | fn~oej: I tested with SIP, IAX, Zap. |
| 09:05.34 | [av]bani | extensions direct into sip are ok |
| 09:05.36 | felix | same problem |
| 09:05.38 | [av]bani | bridged all die |
| 09:05.38 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:05.39 | oej | jpk: More thanks |
| 09:05.42 | jpk | Setting the date back one day fixes it. |
| 09:05.46 | [av]bani | iax/sip dead as a doornail |
| 09:05.51 | fdask | does asterisk die? or just not pass audio like it should |
| 09:05.54 | oej | One-legged calls without bridges seems to work |
| 09:05.59 | jpk | just no audio. |
| 09:06.06 | jpk | fn~oej: ACK |
| 09:06.08 | [av]bani | signaling seems to work, just no audio |
| 09:06.11 | Drew | actually you get about half a sek of audio - then it dies |
| 09:06.11 | jpk | Even Voicemail etc. |
| 09:06.12 | oej | Next question: Does transcoding work? |
| 09:06.15 | [av]bani | it reports ringing and answer and hangup |
| 09:06.19 | jpk | fn~Drew: I second that. |
| 09:06.22 | oej | Will transcoding affect this bug? |
| 09:06.41 | oej | With transcoding, the bridge takes another path |
| 09:06.42 | jpk | oej: I suppose it does since my voiceprompts have to be transcoded AFAIK. And the voiceprompts work. |
| 09:06.50 | Drew | if you look at the RTP debug you will see that a few packets go through - but the stream gets killed as soon as the call is answerd |
| 09:06.53 | oej | voiceprompts is one-legged |
| 09:07.04 | oej | Let's say you have sip/g711 calling SIP/gsm |
| 09:07.18 | oej | is that different from SIP/g711 calling SIP/g711 ? |
| 09:07.25 | jpk | Well. I am pretty sure I tested ZAP/g711 to IAX/gsm |
| 09:07.27 | jpk | no difference. |
| 09:07.38 | jpk | IAX/g711 to SIP/g711 dead as well. |
| 09:07.45 | oej | So all bridges are dead |
| 09:07.48 | jpk | Exactly. |
| 09:07.50 | Drew | i dont think it depends on the codec - its got to do with the briges |
| 09:07.53 | Drew | bridges |
| 09:07.55 | oej | And it can't be a recent patch, since tarballs are affected |
| 09:08.05 | jpk | I do not even see an "attemped" bridge. |
| 09:08.07 | oej | Is it the same for 1.2, 1.2.1 and 1.2.2 |
| 09:08.12 | Drew | yes - i got my tarball last weekend |
| 09:08.22 | oej | Yes, but does it exist in 1.2.0 ? |
| 09:08.28 | jpk | I usually recall some sort of "attempting to bridge x to y" which I do not see anymore. Might be something else though. |
| 09:08.33 | oej | I remember some changes to bridging recently |
| 09:08.42 | Drew | i think its only 1,2,2 |
| 09:08.54 | oej | jpk: Keving changed bridging a few days ago, but that should *not* affect 1.2.0 or 1.2.1 installations |
| 09:08.58 | RoyK | oej: it's indeed a problem also with MoH |
| 09:09.16 | oej | RoyK: On a one-legged call - calling to an extension with musiconhold() ? |
| 09:09.23 | RoyK | yes |
| 09:09.27 | RoyK | i just tried from zap |
| 09:09.31 | *** join/#asterisk jazzplyer (n=jhaar@222-153-249-70.jetstream.xtra.co.nz) |
| 09:09.46 | Drew | but i.e. an echotest works |
| 09:09.51 | jpk | fn~RoyK: That is interesting since echotest works etc. |
| 09:10.09 | RoyK | it does not |
| 09:10.27 | oej | From SIP echotest works |
| 09:10.33 | RoyK | from zap it does not |
| 09:10.54 | felix | I can call MoH from Zap |
| 09:10.58 | Drew | tested SIP echotest - works |
| 09:11.06 | jpk | Ok. Honestly have not tried Zap/echotest, simply assumed it worked. |
| 09:11.53 | *** join/#asterisk Abbas (n=Abbas@203.81.225.61) |
| 09:11.58 | RoyK | neither MoH or echotest works |
| 09:11.58 | fdask | i dont understand how a system date would have anything to do with bridging code? |
| 09:12.00 | [av]bani | sip echotest works here |
| 09:12.22 | Drew | how about timing information? |
| 09:12.35 | felix | any idea what to do to get a running system? I have a call center getting no calls at all :-( |
| 09:12.46 | [av]bani | fdask: packets are timestamped |
| 09:12.59 | [av]bani | felix: roll your clock back |
| 09:13.00 | RoyK | hm |
| 09:13.06 | RoyK | playback seems to work, though |
| 09:13.09 | RoyK | strange.... |
| 09:13.13 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 09:13.31 | oej | bani: Thanks, that's a clue |
| 09:14.05 | [av]bani | sip MoH works here |
| 09:14.15 | [av]bani | internal extension, not bridged |
| 09:14.16 | JunK-Y | saydigits too. |
| 09:15.14 | Drew | but how can bridging affect MoH? |
| 09:15.24 | jpk | Are we sure no funny guy implemented some sort of timebomb? |
| 09:15.33 | oej | jpk: No |
| 09:15.43 | jpk | Thought so. *searching* |
| 09:16.00 | oej | If the MOH has to be transcoded, it may be affected |
| 09:16.02 | RoyK | oej: ding |
| 09:16.06 | RoyK | oej: this is strange |
| 09:16.13 | oej | Damn right |
| 09:16.14 | RoyK | with zaptel, playback works |
| 09:16.31 | Drew | could be difficult to search the code for a timebomb if its not in the plain |
| 09:16.32 | RoyK | and if then calling echo right after, echo seems to work for about a second |
| 09:16.52 | RoyK | oej: try calling +4721973540 |
| 09:16.55 | oej | Let's assume it's an ordinary bug, not a time bomb |
| 09:17.09 | jpk | sure. |
| 09:17.18 | RoyK | oej: it did indeed happen last night |
| 09:17.19 | jpk | Is 1.0.9 affected? |
| 09:17.21 | RoyK | no |
| 09:17.26 | RoyK | at least not 1.0.7 |
| 09:17.31 | jpk | thought so. Very strange. |
| 09:17.34 | RoyK | haven't tried 1.0.9, though |
| 09:17.43 | [av]bani | exten => 500,1,Dial(IAX2/guest@misery.digium.com/s@default) |
| 09:17.45 | [av]bani | no audio |
| 09:17.59 | [av]bani | <PROTECTED> |
| 09:18.02 | [av]bani | <PROTECTED> |
| 09:18.04 | [av]bani | <PROTECTED> |
| 09:18.07 | [av]bani | <PROTECTED> |
| 09:18.09 | [av]bani | <PROTECTED> |
| 09:18.12 | [av]bani | <PROTECTED> |
| 09:18.14 | [av]bani | and thats all she wrote |
| 09:18.18 | oej | RoyK: That number seems to work |
| 09:18.28 | oej | I am getting echo |
| 09:18.40 | RoyK | oej: I only get echo the initial second. |
| 09:18.41 | RoyK | also |
| 09:18.46 | RoyK | i just downgraded to rev 8000 |
| 09:19.00 | Drew | hey it will work agian in the future |
| 09:19.21 | RoyK | but i still get only the initial echo |
| 09:19.23 | Drew | i set my date to 2008 - it works |
| 09:19.30 | RoyK | after a second or two it doesn't work |
| 09:21.12 | fdask | http://forums.digium.com/viewtopic.php?t=4008 |
| 09:21.15 | oej | Can we find that change to briding that was recently done |
| 09:21.52 | [av]bani | Drew: try 1/26/06 |
| 09:23.05 | oej | Can someone try without G711 ? |
| 09:23.05 | RoyK | [av]bani: what revision is that? |
| 09:23.23 | JunK-Y | brb |
| 09:23.25 | [av]bani | RoyK: 1.2.2 |
| 09:23.28 | oej | RoyK: :-) |
| 09:23.48 | [av]bani | i didnt change it, just that you were fiddling with dates, you should try :) |
| 09:24.40 | Drew | ok tryed it - tomorrow doesnt work ;) |
| 09:24.49 | [av]bani | D: |
| 09:24.59 | [av]bani | but 2008 does? :() |
| 09:25.02 | skeffling | Aetting back 1 week on a 1.2.2 system has got them working again. (well, atleast for a week!) |
| 09:25.13 | RoyK | [av]bani: revision, as in 'from svn' not version |
| 09:25.27 | [av]bani | RoyK: whatever rev 1.2.2 release is |
| 09:25.36 | oej | Wonder if compilation with TRACE_FRAMES gives any additional data |
| 09:26.34 | mgoh | may I know who has try running asterisk in ram disk? |
| 09:26.35 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 09:27.06 | Drew | avbani - yes - audio gets killed on *nix timestamp 1138166332 - it goes back on somewhere in Jan 2008 |
| 09:27.24 | Drew | havent found out the other exact timestamp ;) |
| 09:27.26 | RoyK | sounds reasonable..... |
| 09:27.33 | RoyK | 2 years of freedom |
| 09:27.52 | Drew | lol |
| 09:28.46 | fenlander | does someone have a list of what does and what does not work? |
| 09:29.02 | RoyK | fenlander: fill it into mantis as you go |
| 09:29.13 | RoyK | fenlander: that's the right place to place it anyway |
| 09:29.23 | oej | The only timestamp in a frame is the delivery time... |
| 09:29.33 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 09:29.42 | fenlander | RoyK: ok |
| 09:29.51 | *** join/#asterisk shaun-K (n=chatzill@shaun.1ec.aaisp.net.uk) |
| 09:30.30 | fdask | y2k6 bug heh |
| 09:30.32 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:31.15 | RoyK | y2k6m1d25 bug |
| 09:31.39 | RoyK | can't be that bad either. some customers are complaining about noise. how that problem has gone as well...... |
| 09:33.17 | Drew | mantis account creation is brocken |
| 09:33.26 | oej | Hmm. RTP debug shows no incoming audio... |
| 09:33.35 | *** join/#asterisk grexk (n=Server@210.213.176.37) |
| 09:33.48 | RoyK | oej: it's not RTP |
| 09:33.56 | Drew | royk - what do you mean noise? |
| 09:34.08 | RoyK | oej: as I said I had this problem with Zap/IAX2 |
| 09:34.16 | oej | But when I call Asterisk with a SIP phone, I am supposed to see incoming RTP :-) |
| 09:34.27 | heroine | hi ppl |
| 09:34.29 | RoyK | Drew: just customers complaining about jitter every now and then |
| 09:34.33 | Drew | ah ok |
| 09:34.42 | Drew | not bug related |
| 09:34.42 | RoyK | hi, heroine |
| 09:34.45 | oej | I see outgoing RTP with echo or so |
| 09:35.00 | RoyK | wow |
| 09:35.03 | RoyK | blue sky today |
| 09:35.12 | Drew | oej - i hat some RTP from the moment the call came in to the moment when it was answerd |
| 09:35.18 | RoyK | and I'm going to work on the server farm, FAR away from sun |
| 09:35.19 | RoyK | :( |
| 09:35.25 | oej | Test with rtp debug |
| 09:35.36 | Drew | yep thats what i did |
| 09:35.54 | Drew | i had rtp traffic up to answering the call |
| 09:36.20 | Drew | and thats the aprox. 1 sek audio you get when you answer |
| 09:36.26 | mut | well |
| 09:36.30 | mut | the switchover went swimmingly |
| 09:36.42 | heroine | RoyK, that's the same for all of us, staying in front of a monitor and leaving work when the sun has disappeared :/ |
| 09:37.00 | Drew | you could do nightshifts ;-) |
| 09:37.47 | Drew | cool - im getting more bandwith for free *yay to my provider* |
| 09:37.55 | kram | anybody having |
| 09:38.10 | kram | anybody having but #6349? |
| 09:38.48 | oej | kram: Welcome!!!! |
| 09:38.56 | kram | thanks oej |
| 09:38.58 | oej | kram: You are needed! |
| 09:39.05 | kram | brian capouch called me about bug #6349 |
| 09:39.11 | kram | i've uploaded a patch, but i have no noet, only GPRS |
| 09:39.17 | kram | so i need someone to test it |
| 09:39.21 | oej | What was it? |
| 09:39.30 | kram | if it's good, we should get it in |
| 09:39.30 | mut | i can test it |
| 09:39.33 | kram | and then i'll call russell |
| 09:39.34 | oej | A simple patch or a long one |
| 09:39.37 | kram | try to wake him up |
| 09:39.40 | kram | it's on bug #6349 |
| 09:39.43 | *** join/#asterisk Peste (i=Peste@195.230.162.134) |
| 09:39.48 | oej | kram: I am awake and can commit |
| 09:39.56 | Peste | hello everybody |
| 09:40.04 | mutilator | kram |
| 09:40.12 | kram | looks like it wen tin with #7826 |
| 09:40.20 | *** join/#asterisk Givur (n=mail@G9d27.g.pppool.de) |
| 09:40.21 | Givur | Hi all. |
| 09:40.25 | reza | there is a patch out already?! |
| 09:40.35 | reza | for the timebomb problem? |
| 09:40.38 | oej | kram: Ok, this patch for 1.2 branch? |
| 09:40.39 | kram | but please test #6349 |
| 09:40.43 | mutilator | heh timebomb problem |
| 09:40.44 | kram | it should work with either |
| 09:40.53 | [av]bani | compiling |
| 09:40.58 | mutilator | 1.2.1 works fine.. |
| 09:41.03 | kram | i've got no way to test from here |
| 09:41.12 | reza | so what was the problem? |
| 09:41.15 | oej | We have many people online to test |
| 09:41.41 | oej | royk and the rest: Please test patch in bug tracker now |
| 09:41.43 | kram | when we confirm it works i'll explain |
| 09:41.47 | oej | Confirm in this channel |
| 09:41.48 | Peste | can somebody help me? i got a "Unable to create channel of type 'Zap'" error, and i cant fix this. |
| 09:41.52 | [av]bani | fixed |
| 09:42.03 | reza | can someone give me a url to the bug? |
| 09:42.10 | mutilator | http://bugs.digium.com/view.php?id=6349 |
| 09:42.13 | [av]bani | 1.2.2 + mydiff.txt = \o/ |
| 09:42.33 | reza | you got to love opensource |
| 09:42.38 | oej | Peste: Please wait, we're fixing something else now. Thanks. |
| 09:42.39 | kram | what's "\o/ mean? |
| 09:42.43 | reza | bug found, fix out within an hour or two |
| 09:42.50 | Peste | i have a TE110P Card (using E1) linked with an nms card. all modules loaded so the link is green |
| 09:42.57 | viperdude | hi guys anyone know why when someone transfers to another extension it creates 2 CDR entries? |
| 09:43.08 | sm7xab | Hi! Does anyone know if it's possible to detect the number of active calls in the dialplan? Would need this to try and get a queue system working. |
| 09:43.14 | [av]bani | kram: \o/ = http://www.vpaf.uni.edu/unidome/spirit/pictures/cheer%20pom%202.jpg |
| 09:43.36 | [av]bani | @\o/@ if you like |
| 09:43.46 | *** part/#asterisk shaun-K (n=chatzill@shaun.1ec.aaisp.net.uk) |
| 09:44.00 | viperdude | sm7xab: 'show channels' from the CLI/Manager |
| 09:44.01 | Drew | what do i have to do with mydiff.txt to patch * ?? |
| 09:44.09 | kram | oej: since i'm on gprs, can you commit the change to 1.2 please |
| 09:44.09 | kram | i'm trying trunk now |
| 09:44.16 | oej | kram: Sound is back on my system! |
| 09:44.20 | oej | kram: Will do |
| 09:44.23 | oej | kram: And trunk |
| 09:44.27 | kram | great |
| 09:44.31 | mutilator | also works here |
| 09:44.34 | mutilator | zap -> sip |
| 09:44.35 | sm7xab | viperdude: Jupp, that's possible, but the CLI is a bit difficult to use in a dialplan :-( |
| 09:44.39 | kram | i'll try to get a new release out |
| 09:44.42 | kram | brb |
| 09:45.06 | Drew | mutilator - how do you patch * with mydiff.txt? |
| 09:45.18 | razu_ | hi ... does anyone have Sipura Profile compiler to share ? :) |
| 09:45.28 | Peste | <PROTECTED> |
| 09:45.31 | oej | drew: patch < mydiff.txt within the source code directory |
| 09:45.31 | grexk | patch -p1 ../mydiff |
| 09:45.31 | sm7xab | viperdude: I want something like: 1 phone active, dial others short time then put in queue. |
| 09:45.39 | [av]bani | razu_: dont need a compiler, just use xml |
| 09:45.54 | grexk | cd to souce code dir |
| 09:46.28 | mutilator | yea like that |
| 09:46.31 | viperdude | sm7xab you mean you want to dial some extens and if they dont answer put in queue? |
| 09:46.39 | razu_ | [av]bani : k |
| 09:47.14 | Drew | so i cd to /usr/src/asterisk-1.2.2 put the txt file there and run "patch < mydiff.txt" ?? |
| 09:47.22 | sm7xab | viperdude: Yes, bu only if one or two extensions are active. Otherwise I want to send people to voicemail :-) |
| 09:48.09 | viperdude | you would need to write a AGI that logs into Manager and looks up how many active calls and the fork in the dialplan based on the result |
| 09:48.34 | mutilator | so plz to explain when you're done kram |
| 09:48.36 | mutilator | thx |
| 09:48.41 | [av]bani | razu_: "sipura mass deployment" on voip-info.org |
| 09:49.17 | razu_ | [av]bani : thx |
| 09:49.25 | sm7xab | viperdude: Aha! Seems a bit more advanced than I had hoped. Well, now I know where to look. I'll try to fix it when I've got the time... |
| 09:49.33 | grexk | Im just new to asterisk and I want to know more about it. Right now Im reading asterisk documentation and I dont understand terms. Can someone give me hints? |
| 09:49.36 | oej | svn for 1.2 updated with markster's fix to the timebomb bug |
| 09:49.56 | grexk | or related articles to read on for a newbhie |
| 09:50.18 | zoa | oej, what is it about ? |
| 09:50.22 | zoa | this timebomb ? |
| 09:50.25 | zoa | how does it happen ? |
| 09:50.34 | Drew | zoa - http://bugs.digium.com/view.php?id=6349 |
| 09:50.50 | kram | okay apparently in one of the recent diffs that went in, a check was removed for whether we actually had atimeout or not |
| 09:51.00 | oej | zoa: No audio |
| 09:51.23 | zoa | ah :) |
| 09:51.26 | zoa | get it |
| 09:51.28 | oej | kram: And that caused error tonight? |
| 09:51.34 | zoa | im going to look hard at iax2 |
| 09:51.41 | mutilator | why'de it do it at a specific time.. |
| 09:52.00 | reza | had something to do with whatever ast_tvnow() and bridge_end were |
| 09:52.01 | Drew | black magic.. ;) |
| 09:52.21 | kram | oej: yes, that is what caused it |
| 09:52.27 | kram | in a sense we're very lucky |
| 09:52.39 | kram | that bug was very recently introduced (last two weeks) |
| 09:52.46 | zoa | oej, davy is going to call you for the trainings |
| 09:52.55 | oej | zoa: Hotel information on the web site now |
| 09:53.02 | zoa | that woman that called you yesterday was my assistant |
| 09:53.09 | zoa | i forgot to tell her about your email :) |
| 09:53.11 | kram | i can't get a hold of kevin, russell or kenny |
| 09:53.20 | kram | so i can't do the 1.2 release myself |
| 09:55.21 | [av]bani | bugs.digium.com seems dead |
| 09:55.32 | RoyK | hm |
| 09:55.32 | RoyK | yes |
| 09:55.34 | [av]bani | stuck hard |
| 09:55.54 | reza | kram - i'm looking at channel.c off cvs and can't see where the patch is supposed to fit |
| 09:56.01 | *** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.241) |
| 09:56.15 | Drew | yep here two |
| 09:56.16 | Powerkill | hi |
| 09:56.39 | *** join/#asterisk crich1999 (n=crich@p54BFC05F.dip0.t-ipconnect.de) |
| 09:56.51 | Drew | ddos on bugs.digium.com?? |
| 09:56.57 | Powerkill | I have a problem when I call via IAX suddenly since this morning i got IAX2/***** is ringing and when I answer I got IAX2/****** stopped sounds |
| 09:57.17 | Drew | powerkill - http://bugs.digium.com/view.php?id=6349 |
| 09:57.28 | reza | hmm |
| 09:57.50 | reza | is the svn and cvs synched up, or is the cvs no longer being supported? |
| 09:58.16 | kram | cvs takes a long time to get synced up |
| 09:58.36 | [av]bani | bleh who would want to use cvs when you can use svn |
| 09:58.56 | reza | I did a cvs annotate to look for the recent changes, and the latest dates are back in October |
| 09:59.01 | crich1999 | kram, you're still up and fixing issues over GPRS ? |
| 09:59.04 | crich1999 | <PROTECTED> |
| 09:59.15 | reza | i tried using svn before, but the server gave me a 404 |
| 09:59.24 | reza | svn.digium.com/svn or something like that? |
| 09:59.31 | kram | crich: oej got it in there for me |
| 09:59.38 | kram | oej: did you already put it in both? |
| 09:59.45 | crich1999 | kram, he did |
| 09:59.54 | kram | what commit message? :) |
| 09:59.57 | oej | kram: yes, it is in both. Don't know when it will be copied from origsvn to svn |
| 10:00.06 | oej | kram: You'll find out :-) |
| 10:00.23 | oej | kram: Issue #6439 - the timebomb bug |
| 10:00.37 | oej | Or rather 6349 |
| 10:00.49 | oej | I mailed to the -users and -dev mailing list |
| 10:01.08 | oej | I can't change the topic here in the channel |
| 10:01.16 | *** join/#asterisk zzxxcc (n=zzxxcc@221.232.13.178) |
| 10:01.25 | crich1999 | I assume this issue exist in 1.2.1 and 1.2.2 also, will they get the fix ? |
| 10:01.35 | mutilator | it doesn't exist in 1.2.1 |
| 10:01.39 | [av]bani | i predict a 1.2.3 release |
| 10:01.43 | mutilator | atleast i'm running it w/o the problem |
| 10:01.52 | clive- | any news on a sip jitter buffer? |
| 10:02.02 | oej | The fix will be in the 1.2.3 release |
| 10:02.03 | crich1999 | but it exsits for sure in 1.2.2 |
| 10:02.11 | mutilator | yes, it doesin 1.2.2 |
| 10:02.15 | [av]bani | clive -> beta one, check mantis |
| 10:02.21 | oej | Mark is trying to find a release engineer to release a new release |
| 10:02.21 | clive- | thanks |
| 10:02.26 | reza | so is it in svn, or some svn staging server? and what's the svn url? |
| 10:02.34 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:02.46 | oej | reza: http://www.asterisk.org will tell you more about URLs for SVN |
| 10:03.11 | oej | reza: Or you can simply take the patch from issue 6349 and apply it to your source code |
| 10:03.28 | reza | oej - my source was from cvs which didnt match up |
| 10:03.58 | zoa | clive the jitter buffer exists and works |
| 10:04.02 | zoa | just patch your asterisk for it |
| 10:04.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 10:04.51 | dudes | I remember while ago I got head and it was in the furture, so I didn't get any errors which. In any event, I quickly realized that it's cause they hadn't been invented yet. |
| 10:04.55 | oej | reza: Ok |
| 10:05.39 | Powerkill | Drew thanks, but why this happens today ? |
| 10:05.58 | [av]bani | mantis recovered |
| 10:06.10 | zzxxcc | hi, all |
| 10:06.25 | zzxxcc | What is Steve Underwood's nickname here? |
| 10:06.31 | zoa | coppice |
| 10:06.37 | zzxxcc | zoa:: thx |
| 10:06.40 | clive- | zoa thanks |
| 10:06.47 | Givur | I need to say, that was really fast :) Good work from the Asterisk team. |
| 10:06.56 | zoa | i think the bug number is 3854 or so |
| 10:06.58 | Givur | (or better the coders) |
| 10:07.11 | oej | Givur: Actually, Mark :-) |
| 10:07.36 | zoa | Mark for president |
| 10:07.49 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 10:08.05 | kram | thanks |
| 10:08.06 | Drew | ~ 6h bug reponse time - i wish M$ and other large companys were as fast as that... |
| 10:08.13 | kram | i appreciate the vote of confidence |
| 10:08.25 | MrChimpy | "I remember while ago I got head and it was in the furture" |
| 10:08.33 | oej | More karma points for Mark .-) |
| 10:09.31 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 10:09.33 | reza | so today is my frist day w/ asterisk and i got hit by the bug, but reverting to an older version didnt work (using NuFone to dial out). It dials, but no audio. Is there anything else I can try to see if the problem is my side or if it's Nufone's problem? |
| 10:09.47 | [av]bani | Drew: well if youre ever disappointed with * response time you can demand a full refund of the purchase price |
| 10:09.54 | oej | nufone may be asleep and not patching their servers |
| 10:10.04 | mutilator | heh |
| 10:10.11 | mutilator | i was supposed to roll out a new server today |
| 10:10.12 | Drew | lol |
| 10:10.23 | mutilator | guess i picked a good day eh |
| 10:10.34 | reza | so the problem could be on their side -- has anyone recently used a provider that works? |
| 10:10.50 | [av]bani | reza: nat? |
| 10:10.53 | Drew | my proivder works - but they dont use * |
| 10:11.07 | reza | i can call my cell, and it works, so i dont think it's the nat |
| 10:11.22 | mutilator | yikes |
| 10:11.25 | Stephnie | http://pastebin.ca/38503 <--- any one please help |
| 10:11.26 | mutilator | 351mb voicemail |
| 10:11.29 | reza | well, it works in that the cell rings, but when i pick up, no audio |
| 10:12.59 | oej | bug tracker is slow now |
| 10:13.50 | zzxxcc | seems that coppice is not online. When could I expect he be here? |
| 10:13.58 | zoa | ~seen coppice |
| 10:14.01 | jbot | coppice <n=chatzill@193.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 19h 6m 3s ago, saying: 'relativeambition: it does not'. |
| 10:14.03 | *** join/#asterisk diLLec (n=dillec@a15182648.alturo-server.de) |
| 10:14.12 | zoa | probably in a few hours |
| 10:14.38 | diLLec | hey |
| 10:14.57 | Stephnie | <PROTECTED> |
| 10:15.00 | Stephnie | :) |
| 10:15.32 | *** join/#asterisk fulgas (n=fulgas@209.8.233.12) |
| 10:15.39 | reza | where's the first place to look regarding jittery audio - including the phone ringing sound |
| 10:15.56 | Drew | reza - how is your bandwith? |
| 10:16.07 | zoa | jitter, timing device, timing frequency in kernel |
| 10:16.11 | *** join/#asterisk SrFr (n=blaat@cust.12.229.adsl.cistron.nl) |
| 10:16.31 | reza | 3mbit down, 512 up |
| 10:16.34 | Drew | it could also be an overloaded internet pipe |
| 10:16.42 | reza | no net traffic |
| 10:16.49 | reza | i think it's my side |
| 10:17.03 | reza | doesn't my server generate the "ringing" sound I hear on the phone? |
| 10:17.03 | Drew | ok then forget the pipe - look at what zoa mentioned... |
| 10:17.31 | Drew | i think that depends |
| 10:17.35 | dudes | reza - what's your ping like to your provider? |
| 10:17.55 | reza | rtt min/avg/max/mdev = 56.277/57.321/57.868/0.698 ms, pipe 2 |
| 10:18.20 | reza | it doesnt always happen |
| 10:18.50 | reza | sometimes it will ring fine once or twice, then it becomes jittery |
| 10:18.50 | zoa | its not your network probably |
| 10:18.52 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 10:19.05 | dudes | iax/sip? |
| 10:19.18 | reza | i am sharing an irq w/ usb device, but there is no usb device plugged into anything |
| 10:19.22 | reza | iax to NuFone |
| 10:19.32 | reza | <PROTECTED> |
| 10:19.33 | Drew | ping to my provider is about 25ms |
| 10:19.41 | dudes | What's your jitter bugger |
| 10:19.48 | dudes | err buffer settings like |
| 10:20.01 | Stephnie | any one good in PERL?? |
| 10:20.27 | oej | brookshire: Online? |
| 10:21.00 | reza | jitterbuffer=no |
| 10:21.00 | reza | forcejitterbuffer=no |
| 10:21.12 | reza | stephnie - what's your question? |
| 10:21.40 | Stephnie | plz check it out http://pastebin.ca/38503 |
| 10:22.08 | diLLec | anyone can help with aborting SIP RTP Streams after 1 or 2 seconds ? |
| 10:22.28 | dudes | Drew - rtt min/avg/max/mdev = 1.051/1.483/2.050/0.378 ms <--- Straight to Commpartners |
| 10:22.46 | Drew | lol - what pipe have you got? |
| 10:22.53 | Drew | E3 or something? |
| 10:22.57 | *** join/#asterisk lancey (i=Shady@support.net1.cc) |
| 10:23.00 | lancey | hi guys |
| 10:23.01 | Drew | dillic - get this patch - http://bugs.digium.com/view.php?id=6349 |
| 10:23.01 | dudes | It's a server Carrier One |
| 10:23.02 | lancey | i' |
| 10:23.09 | dudes | err at Carrier One |
| 10:23.11 | lancey | i'm having a strange problem with asterisk on fbsd 6.0 |
| 10:23.23 | lancey | there's no sound after the call is answered, anyone ran into this? |
| 10:23.24 | Drew | lancey - let me guess - http://bugs.digium.com/view.php?id=6349 |
| 10:23.41 | Drew | could somebody change the channel topic?? |
| 10:23.45 | lancey | Drew thank you! |
| 10:23.50 | diLLec | thats not true, ain't it ?! |
| 10:24.23 | diLLec | i rarley sit three or more hours to find any solution ;) |
| 10:24.38 | Drew | ;) |
| 10:24.38 | lancey | it says it's solved in SVN? |
| 10:24.40 | reza | stephnie - what is your question? what do you want to do? |
| 10:24.43 | oej | Drew: I tried changing the topic, did not work - only a few people can |
| 10:24.46 | lancey | i did a checkout an hour ago, my bad :/ |
| 10:25.13 | Stephnie | reza: did you check my question at pastebin? |
| 10:25.43 | Drew | oej - if you know somebody who can - that would be greate |
| 10:25.49 | Drew | -e |
| 10:25.59 | reza | looks ok, though there isn't much going on |
| 10:26.19 | diLLec | is there any reason why the problem occurs ?= |
| 10:26.23 | reza | well |
| 10:26.26 | reza | actually yeah |
| 10:26.33 | reza | you use $dbh before you define it |
| 10:26.40 | reza | so get rid of the disconnect line |
| 10:26.43 | reza | you dont need it anyhow |
| 10:27.20 | Stephnie | reza: $dbh = DBI->connect($dsn, 'sa', 'passwd'); <---when I comment out this line... then next line SAY NUMBER WORKS .....otherwise it doesnt.. |
| 10:27.43 | reza | get rid of the disconnect line |
| 10:28.00 | Stephnie | ok going to comment it out and check the script |
| 10:28.00 | reza | run it with the -w flag |
| 10:28.16 | reza | so anyone have any other hints about the jitter? |
| 10:28.30 | Drew | dillec - look at the patch ;) |
| 10:28.59 | reza | actually, is jitter the right term? |
| 10:29.06 | Drew | yay - my * has finished compiling the patch |
| 10:29.14 | reza | the audio has breaks in it |
| 10:29.24 | reza | it stops for a fraction of a second then starts again |
| 10:29.26 | reza | is that jitter? |
| 10:29.43 | Stephnie | reza : I have commented out $dbh->disconnect if $dbh; .....and -w ....but still dont hear anything from SAY NUMBER |
| 10:30.33 | mutilator | what in the frickin hell |
| 10:30.34 | JunK-Y | theres a bug with audio at this time. |
| 10:30.41 | mutilator | someone in Albuqurque, NM |
| 10:30.47 | mutilator | is dialing one number after another |
| 10:30.53 | mutilator | and there is no one even registered for those |
| 10:31.00 | reza | it's having issues connection; surprised it doesn't print out an error |
| 10:31.12 | Drew | mutilator - maybe a phreaker |
| 10:31.26 | mutilator | so far it's gone 269-310-2499 down to 2464 and keeps going down |
| 10:31.33 | lancey | my * is from 2006-01-22, and there's also a problem with the audio |
| 10:31.40 | Drew | try calling him ;) |
| 10:31.48 | JunK-Y | i know it affects all versions apparently. |
| 10:31.49 | lancey | after bridging there's audio for a second or so |
| 10:31.57 | JunK-Y | or a lot of these versions. |
| 10:32.12 | mutilator | it's not even the same callerid calling me tho |
| 10:32.14 | Stephnie | reza: should it print out errors at asterisk CLI? |
| 10:32.21 | reza | no |
| 10:32.23 | Drew | lancey - http://bugs.digium.com/view.php?id=6349 |
| 10:32.28 | reza | in perl |
| 10:32.43 | lancey | Drew: it says "from jan 25 2006" |
| 10:32.54 | mutilator | it's like non sensical |
| 10:33.01 | mutilator | 5058843519 |
| 10:33.03 | Drew | yes - its not the build - its the date |
| 10:33.06 | lancey | aaah |
| 10:33.08 | lancey | :/ |
| 10:33.09 | mutilator | 5058830278 |
| 10:33.18 | mutilator | 505-883-0278 |
| 10:33.19 | mutilator | er |
| 10:33.24 | mutilator | 5058727811 |
| 10:33.24 | JunK-Y | just set it to 2008. |
| 10:33.25 | lancey | holly shit |
| 10:33.35 | Drew | you can spoof your caller ID |
| 10:33.42 | lancey | are all asterisk versions affected? |
| 10:33.53 | reza | 1.2.2 |
| 10:34.06 | mutilator | why bother starting them all with 505 tho? |
| 10:34.09 | JunK-Y | lancey: a lot. |
| 10:34.34 | reza | junky - actually, but was only introduced 2 weeks ago or so |
| 10:34.41 | mutilator | well |
| 10:34.41 | Drew | mut - no idea - trace the call ;) |
| 10:34.47 | mutilator | they hit below my registered number |
| 10:34.49 | JunK-Y | it was introduced at midnight |
| 10:34.53 | mutilator | i only have a block of 50 there... |
| 10:35.00 | mutilator | o well |
| 10:35.15 | reza | i cant msg you back |
| 10:35.21 | reza | stupid server |
| 10:35.22 | reza | umm |
| 10:35.36 | reza | are you sure the server is working and you have no firewall issues? |
| 10:35.36 | lancey | JunK-Y the mantis says it's fixed in latest SVN |
| 10:35.52 | JunK-Y | i tried it 2 hours ago and it was not. |
| 10:35.56 | lancey | me too |
| 10:36.01 | lancey | i'm now checking it out again |
| 10:36.05 | reza | it's fixed as of 30 minutes ago or so |
| 10:36.10 | Drew | junky - just patch it with the txt file on mantis |
| 10:36.20 | reza | (in svn) |
| 10:36.51 | JunK-Y | ha, i was partying with etel guys. |
| 10:36.56 | dudes | So after Jan 25 you won't get sound out of SIP? Or is it before, Nice |
| 10:37.07 | reza | after |
| 10:38.43 | *** join/#asterisk ctm (n=ctm@dsl-220-253-74-87.NSW.netspace.net.au) |
| 10:39.27 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 10:39.31 | *** part/#asterisk ctm (n=ctm@dsl-220-253-74-87.NSW.netspace.net.au) |
| 10:40.23 | *** join/#asterisk ctm64 (n=ctm@dsl-220-253-74-87.NSW.netspace.net.au) |
| 10:40.47 | MikeJ__ | hello |
| 10:41.29 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com) |
| 10:41.57 | Powerkill | somebody can explain why this bug come today ? |
| 10:42.03 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 10:42.50 | dudes | Cause head came from the furture |
| 10:42.55 | dudes | err future |
| 10:43.55 | mutilator | anyone ever used copper mountain dlsams? |
| 10:43.57 | mutilator | dslams |
| 10:44.35 | Drew | powerkill - this was on the channel: |
| 10:44.38 | Drew | <kram> okay apparently in one of the recent diffs that went in, a check was removed for whether we actually had atimeout or not |
| 10:44.38 | Drew | <oej> kram: And that caused error tonight? |
| 10:44.38 | Drew | <reza> had something to do with whatever ast_tvnow() and bridge_end were |
| 10:44.38 | Drew | <kram> oej: yes, that is what caused it |
| 10:44.38 | Drew | <kram> in a sense we're very lucky |
| 10:44.39 | Drew | <kram> that bug was very recently introduced (last two weeks) |
| 10:45.37 | reza | ok, this is pissing me off. it happens on 1 of 5 calls or so. the ringing has massive jitter - it stops and starts like a drum beat |
| 10:45.51 | reza | where can i look for a fix? |
| 10:45.55 | dudes | reza - what codec? |
| 10:46.01 | reza | gsm |
| 10:46.08 | reza | but it's just the ringing sound on my phone so far |
| 10:46.11 | dudes | What proc? |
| 10:46.28 | Drew | is this on all connections internal and external? |
| 10:46.37 | reza | proc? model name : AMD Sempron(tm) Processor 2800+ |
| 10:47.04 | reza | this is with an FSX card (tdm400) making a call out via aix |
| 10:47.27 | reza | i removed the other drivers for the usb stuff so it's on it's own irq |
| 10:47.33 | dudes | reza - Shouldn't be the ponies, heh |
| 10:47.40 | reza | ponies? |
| 10:47.46 | dudes | the processor |
| 10:47.48 | *** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 10:47.59 | reza | ah ;) |
| 10:48.00 | dudes | that should be way more than enough power |
| 10:48.01 | reza | er |
| 10:48.02 | reza | :) |
| 10:48.38 | reza | it's like 'ra ra ra ra' rather than 'riiiiing' |
| 10:48.47 | reza | only sometimes |
| 10:49.03 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 10:49.30 | dudes | If the call itself isn't doing it I'd probably ignore it |
| 10:49.48 | dudes | But that's just me, but I like having a MOH playing while I dial |
| 10:50.29 | reza | they both are |
| 10:50.45 | reza | i got some recording once, and that had jitter in it too |
| 10:51.39 | dudes | I don't know. I haven't had that issue in along time. |
| 10:51.59 | dudes | And I don't really know what resolved it (aside from turning jitter off). |
| 10:52.08 | dudes | and my ISP not being gay |
| 10:52.08 | *** part/#asterisk grexk (n=Server@210.213.176.37) |
| 10:53.02 | reza | i'm killing all unncessesary processes |
| 10:53.21 | dudes | I highly doubt it's a resource issue |
| 10:53.22 | reza | i can dial out, but no audio still - think NuFone needs to upgrade their shit in the morning |
| 10:53.35 | reza | think the problem is with the hardware? |
| 10:53.47 | dudes | What hardware |
| 10:53.53 | reza | the motherboard? |
| 10:53.58 | reza | the version of the tdm card i have |
| 10:54.06 | dudes | What version of tdm? |
| 10:54.09 | reza | why else would it be unfixable? |
| 10:54.20 | reza | doi |
| 10:54.20 | dudes | the jitter |
| 10:54.21 | reza | sorry |
| 10:54.29 | reza | misread resource as resolveable |
| 10:54.31 | reza | it's late |
| 10:54.38 | dudes | I know the feeling |
| 10:54.46 | dudes | it is 5am almost |
| 10:54.55 | *** part/#asterisk Falle (i=falstaf@213.141.80.88) |
| 10:56.12 | Drew | dudes - are you in india or something? *calculating timezones* |
| 10:56.22 | dudes | Minnesota |
| 10:57.02 | dudes | That's a good ways away from India, heh. |
| 10:57.29 | Drew | ups - is west... |
| 10:57.42 | Drew | minus UTC is west |
| 10:58.16 | Drew | its noon here - but ive been up since yesterday... so i know the feeling ;) |
| 11:02.11 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:03.26 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:04.29 | zoa | hey ho olle |
| 11:04.29 | zoa | <PROTECTED> |
| 11:06.08 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 11:06.19 | reza | 3am here |
| 11:07.10 | Drew | pacific coast in that case... |
| 11:08.51 | mutilator | grrrr |
| 11:08.55 | mutilator | this dslam is pissin me off |
| 11:09.09 | mutilator | i have customers on it and an ata on that |
| 11:09.16 | mutilator | dsl modem is just a bridge |
| 11:09.16 | *** join/#asterisk ErMeS|Work (n=ermsewrk@217.220.121.62) |
| 11:09.31 | mutilator | when i load the admin config for the sipura ata the page doesn't fully load |
| 11:09.55 | mutilator | customers also have trouble loading 'random'(same everytime) websites |
| 11:10.17 | Drew | what has a DSLAM got to do with a ATA? |
| 11:10.30 | mutilator | the dslam is the cause of the problem |
| 11:10.45 | mutilator | every customer acts the same way |
| 11:10.53 | mutilator | i have 5 or 6 ata's off that dslam |
| 11:10.57 | mutilator | and they all do the same |
| 11:11.12 | mutilator | the info page shows fine, but go to configure anything and those pages won't load |
| 11:11.23 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:11.45 | mutilator | sounds like an MTU problem.. but mtu is set 1500 all the way through, except the dslam, i can;'t find an mtu setting on that |
| 11:12.05 | mutilator | also tried limiting 10mbit half duplex |
| 11:12.07 | Drew | sorry cant help - i only come in contact with DSLAMs on the 'client site' ;-) |
| 11:12.43 | *** join/#asterisk heroine (n=lahaine@71.68.119-80.rev.gaoland.net) |
| 11:16.03 | *** part/#asterisk ctm64 (n=ctm@dsl-220-253-74-87.NSW.netspace.net.au) |
| 11:17.17 | lancey | mutilator: i had the same problem with pap2-nas |
| 11:17.26 | lancey | it appeared to be MTU |
| 11:17.48 | lancey | i was totally unable to do anything on the web configuration |
| 11:20.48 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 11:21.13 | many | which sip codec is most lossless? alaw? |
| 11:23.18 | diLLec | thats it |
| 11:23.40 | lancey | sip codec? :) |
| 11:23.49 | *** join/#asterisk lahaine (n=lahaine@71.68.119-80.rev.gaoland.net) |
| 11:23.53 | Peste | hello! can somebody help? i have a problem with compiling asterisk 1.2 on a x86_64 machine |
| 11:24.04 | many | Yes. |
| 11:24.31 | many | well, codec in RTP when using SIP as Initiation protocol |
| 11:25.22 | Peste | these are the last line, when i do "make" |
| 11:25.23 | Peste | http://pastebin.com/pastebin.php?dl=522112 |
| 11:26.22 | *** join/#asterisk robin_sz (n=yeah@host-212-18-247-190.static.mailbox.co.uk) |
| 11:26.48 | robin_sz | OK,.... so * ran for a few days, then crapped itself ... can;t see anything in the logs to suggest why .. any other clues? |
| 11:28.02 | robin_sz | any suggestions for ways to automagicvally restart it if it crashes? |
| 11:29.28 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 11:32.32 | qzxcd | robin, perhaps supervise could fit ? |
| 11:32.35 | qzxcd | or svscan ? |
| 11:32.52 | qzxcd | i'm not sure, the tool provided with qmail/djbdns to automatically restart services |
| 11:34.15 | RaYmAn-Bx | I belive the safe_asterisk script automatically restarts asterisk |
| 11:35.37 | mutilator | lancey? |
| 11:35.51 | mutilator | ya fixed it? |
| 11:36.54 | *** join/#asterisk [chico] (n=chico@p54916FF3.dip.t-dialin.net) |
| 11:37.10 | Peste | can somebody help me comiling asterisk? |
| 11:38.45 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 11:39.04 | oej | peste: What is the problem? |
| 11:39.05 | fourcheeze | how do I stop people being able to sip proxy anonymously through * ? |
| 11:39.13 | [chico] | can somebody help me Jan 25 12:38:41 WARNING[17924]: chan_sip.c:12590 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
| 11:39.24 | [chico] | what´s my problem? |
| 11:39.27 | [chico] | please |
| 11:39.29 | oej | fourcheeze: Read the sip.conf sample, there's an option for that |
| 11:39.39 | AndyCap | [chico]: you already have some program listening on 5060? |
| 11:39.40 | fourcheeze | oej: damn must have missed it, thanks |
| 11:40.31 | [chico] | thank you andy |
| 11:40.42 | oej | peste: You do need termcap and ncurses to compile asterisk. Please read the documentation that follows Asterisk within the source code to understand requirements on third party libraries. It's all there |
| 11:40.45 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 11:40.58 | AndyCap | [chico]: netstat -nap |grep 5060 as root may give a clue |
| 11:41.23 | fourcheeze | oej: do you mean "allowguest" |
| 11:41.29 | oej | Right |
| 11:41.32 | Peste | oeg: thanks, but i think i allready got it |
| 11:41.33 | fourcheeze | oej: will that stop people being able to make sip calls inwards? |
| 11:41.46 | yxa | help pls. Any calls made to a zap channel will hangup automatically when the other party picks up |
| 11:41.53 | oej | Fourcheeze: You have to understand how the SIp channel handles incoming calls, let me explain |
| 11:41.59 | fourcheeze | please :-) |
| 11:42.05 | oej | Fourcheeze: First we try to match with an user on the From user name |
| 11:42.11 | oej | Then we match on peer by IP |
| 11:42.40 | oej | If no matches are found and allowguest is on, we send the calls to the context declared in [general] or the "default" context |
| 11:42.56 | oej | If allowguest is off, we deny everything that we can not match with a peer or user definition |
| 11:42.59 | oej | Ok? |
| 11:43.30 | *** join/#asterisk BugKham (n=lamer@gb.ja.95.195.revip.asianet.co.th) |
| 11:43.39 | fourcheeze | oej: very clear, thanks |
| 11:43.44 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 11:43.47 | BugKham | is 1.2.2 really stable? |
| 11:43.52 | oej | fourcheeze: Your welcome |
| 11:43.52 | kink0 | hello |
| 11:44.14 | yxa | dont mean to repeat. Any calls made to a zap channel will hangup automatically when the other party picks up. anyone? |
| 11:44.29 | kink0 | I have a problem with RED alarm ussing TE405 digium card, can somebody give me few help ? |
| 11:44.56 | BugKham | it really gave me headache |
| 11:45.01 | fourcheeze | oej: I understand that when clients subscribe to the proxy they take a session length of 3600 seconds - do you know if there's a way to shorten that? |
| 11:45.44 | oej | There are some confusing options in sip.conf about expiry - I never remember which one is used for what. I may need to sort that out once and for all :-) |
| 11:45.58 | Peste | ya, i relly missed termcap.. thx a lot |
| 11:46.04 | oej | Some of them are for setting what Asterisk accepts for inbound registrations and one is what we ask for in outbound registrations |
| 11:46.14 | Peste | but i still get an error |
| 11:46.31 | fourcheeze | oej: I can find the one to alter the length of registrations |
| 11:46.34 | fourcheeze | but not subscriptions |
| 11:46.35 | Drew | yxa is only the zap channel affected? |
| 11:46.50 | yxa | Drew yeah |
| 11:46.56 | AndyCap | Stupid beginner question: When I place a call through * from iaxcomm to visdn it will stuff up the progress info when I use u-law. (call is connected, but * still plays ringtone) but when I use gsm from iaxcomm -> * it works correctly. Anyone have a clue what to look for? |
| 11:47.18 | oej | Ah, you mislead my by saying "proxy". Since asterisk is not a proxy, I misunderstood you and made a fool of myself. |
| 11:47.32 | oej | Fourcheeze: For subscriptions, I believe we use the same settings as for registrations |
| 11:47.54 | oej | maxepiry, minexpiry |
| 11:47.58 | fourcheeze | asterisk certainly seems to be acting like a proxy |
| 11:48.15 | fourcheeze | I seem to be able to use it as an outbound proxy without registering |
| 11:48.41 | oej | fourcheeze: Yes, we are a SIP registrar/location server, an outbound proxy but *not* a proxy. We never ever forward SIP messages untouched, like a proxy does |
| 11:48.42 | fourcheeze | and maxexpiry doesn't seem to lower the length of the subscription |
| 11:48.49 | oej | It is very different |
| 11:48.54 | fourcheeze | ok |
| 11:48.59 | fourcheeze | I'm trying to do two things |
| 11:49.09 | fourcheeze | 1) Stop people being able to get free calls to our outbound sip route |
| 11:49.25 | fourcheeze | (just by using us as an outbound proxy if they found the ip address of the outbound server) |
| 11:49.27 | oej | for 1) you solve that in the dial plan |
| 11:49.54 | fourcheeze | well no because I can bounce off the outbound proxy avoiding the dialplan altogether |
| 11:50.00 | oej | Because you may want some people using SIP to call you, but not use your Asterisk for outbound services without paying |
| 11:50.16 | fourcheeze | ok, here's how I'm testing |
| 11:50.27 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 11:50.28 | fourcheeze | I have a client (snom phone) that is not registered |
| 11:50.29 | oej | Reading the source, subscriptions use the man/mix expiry settings |
| 11:50.50 | oej | So by lowering the max expiry, you *should* be able to shorten subscriptions |
| 11:51.04 | fourcheeze | I set up function key to call 001234567@outbound.sip.trunk.ip |
| 11:51.26 | fourcheeze | where 001234567 is a valid number to call |
| 11:51.49 | fourcheeze | the outbound provider believes that the call comes from us so allows it |
| 11:51.50 | oej | fourcheeze: You can match that user on From: user name in a user section |
| 11:51.54 | fourcheeze | even though nothing is registered with us |
| 11:52.07 | oej | ok |
| 11:52.08 | AndyCap | fourcheeze: uh, so your provider is faulty? |
| 11:52.19 | fourcheeze | well, I'm guessing so, but I'd like not to rely on them |
| 11:52.37 | fourcheeze | they except everything from particular IP numbers |
| 11:52.41 | fourcheeze | accept even |
| 11:52.55 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 11:52.57 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:52.58 | oej | Gosh, we are only using maxexpiry if the subscription is of one out of three types! |
| 11:53.06 | oej | Yahoo... That has to be fixed. |
| 11:53.19 | fourcheeze | I would very much like it to be fixed |
| 11:53.28 | oej | Fourcheeze: You were right if you're not using a SNOM or so |
| 11:53.42 | fourcheeze | because it seems my snoms always subscribe for 3600 seconds |
| 11:54.06 | oej | Hmmm. SNOMS subscribe for DIALOG_INFO_XML, so then we should apply max expiry |
| 11:54.19 | fourcheeze | hmm |
| 11:54.25 | oej | My Xten Eyebeam use a different format and we do not apply it then |
| 11:54.27 | fourcheeze | ok, I'll have a play with that |
| 11:54.27 | oej | Which is bad |
| 11:54.34 | fourcheeze | however |
| 11:54.43 | fourcheeze | how to stop my outgoing call problem |
| 11:55.18 | fourcheeze | going to see if allowguest=no helps |
| 11:55.48 | oej | fourcheeze: If you configure that phone to use your asterisk as outbound proxy, all calls regardless of URI, will be sent to Asterisk. The phone will stop sending calls directly to your provider |
| 11:56.06 | kink0 | after fews changes and trys I still getting RED alarm for my TE405 card , I am ussing E1 and connected to 2N gateway |
| 11:56.32 | fourcheeze | oej: I'm not making a "call" as such to a provider, I'm entering a sip url |
| 11:56.40 | *** join/#asterisk laichzeit (n=immortal@dsl-165-213-116.telkomadsl.co.za) |
| 11:56.41 | MrChimpy | kink0: sure the cable is right? |
| 11:56.58 | laichzeit | hi anyone have problems installing Asterisk@home 2.3? |
| 11:57.06 | fourcheeze | what seems to happen is that * sees the sip url and proxies straight to the destination of that |
| 11:57.27 | MrChimpy | kink0: should be straight through or a ISDN crossover (not the same as ether xover cable) |
| 11:57.30 | *** join/#asterisk lahaine (n=lahaine@47.68.119-80.rev.gaoland.net) |
| 11:57.31 | kink0 | MrChimpy: I have ACTIVE the interface, I also changed the cable, and ensure to configure NT/M on the 2N side |
| 11:58.07 | kink0 | MrChimpy: may be that, I used a not cross, and now I am ussing a crossover LIKE ether cable. |
| 11:58.56 | kink0 | MrChimpy: is just to cross 2 <-> 3 , 3 <->2, 4 <->5 , 5 <-> 4 ? |
| 11:59.10 | oej | fouorcheeze: If you give a sip phone a URI in which the domain part point somewhere else than to the proxy, it will send the call there when you press the button |
| 11:59.19 | oej | In SIP, the phone rules the network |
| 11:59.53 | oej | If you add a configuration for an outbound proxy in the phone, it will stop trying to resolve the domain part of the URI and send everything to the outbound proxy |
| 12:00.32 | Peste | ok.. i compiled asterisk but got an error at startup: asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_watchdog.so: undefined symbol: ast_load |
| 12:00.43 | Drew | is "#1" a valid extension name? |
| 12:01.02 | Drew | because for some reason it doesnt seam to work |
| 12:01.05 | MrChimpy | kink0: http://www.chebucto.ns.ca/Chebucto/Technical/Manuals/Max/max6000/gs/cablesa.gif |
| 12:01.31 | oej | fourcheeze: Asterisk svn trunk now applies maxexpiry and minexpiry to all subscriptions |
| 12:01.37 | fourcheeze | :-) |
| 12:01.38 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 12:01.42 | jozsab1 | Hy all |
| 12:02.01 | jozsab1 | WARNING[31704]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 12:02.27 | jozsab1 | is there a way to make asterisk work DTMF=inband and using codec g729 ? |
| 12:02.55 | fourcheeze | oej: yes what seems to happen is that the phone *subscribes* to my * server if I put it as an outbound proxy. Then if I dial a sip url all calls go out through the server. That way I can spoof our outbound provider. I'm going to try allowguest=no now to see if that stops it. |
| 12:04.13 | oej | I need to see a SIP debug to understand what's happening |
| 12:04.19 | oej | :-) |
| 12:04.29 | fourcheeze | what's not clear? |
| 12:05.07 | fourcheeze | if I leave out the outbound proxy the call attempts to go straight to the outbound provider. Because it doesn't know about the IP address of the snom it rejects the call (as it should) |
| 12:05.26 | Delvar | jozsab1: no... g729 will distort the tons and they will not be recognised |
| 12:05.30 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 12:05.52 | jozsab1 | Thanks delvar |
| 12:06.12 | jozsab1 | you saved me alot of time and work |
| 12:06.44 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.77) |
| 12:06.59 | Peste | so what can i do? [res_watchdog.so] => (Watchdog Resource) |
| 12:06.59 | Peste | asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_watchdog.so: undefined symbol: ast_load |
| 12:08.11 | Grubs | Oh noes ;) Struck by the dead audio bug! 4 hours gone but fixed with the patch now... hooray! |
| 12:08.37 | Modcuts | morning, How do you make sure your trunks are using the g729 codec if you have registered it in asterisk and you are using it with the hardphone? |
| 12:09.06 | Mavvie | We had a scary situation today with RTP streams. |
| 12:09.15 | Grubs | telling me it was scary! |
| 12:09.25 | Mavvie | As long as the call wasn't established, the RTP stream was there. |
| 12:09.34 | Modcuts | sorted it being stupid |
| 12:09.41 | Mavvie | But the moment asterisk said "call established", it died. |
| 12:10.01 | Grubs | I'm compiling in the fix now |
| 12:10.10 | Mavvie | on network level you saw RTP traffic from the SIP phone, but nothing from the asterisk server. |
| 12:10.23 | oej | fourcheeze: What about removing the domain part of the URI and fix that in the dialplan in the Asterisk server instead, so that the URI on the phone points to asterisk and asterisk handles the call routing? |
| 12:10.29 | Mavvie | rtp debug showed traffic before the call was establish, but afterwards nothing. |
| 12:10.37 | Mavvie | rebooting didn't work. |
| 12:10.45 | Mavvie | going back to 1.2.1 code did work. |
| 12:10.47 | Mavvie | very strange. |
| 12:10.54 | Grubs | oej - thanks for the post with the RTP fix |
| 12:11.19 | Grubs | Mavvie - this one? http://www.asterisk.org/node/55 |
| 12:12.02 | Mavvie | Grubs: W...T...F.... |
| 12:12.11 | Mavvie | what a strange bug. |
| 12:13.23 | *** join/#asterisk john___ (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 12:13.35 | Mavvie | Grubs: any idea what went wrong here? |
| 12:13.37 | *** join/#asterisk coppice (n=chatzill@223.143.17.210.dyn.pacific.net.hk) |
| 12:14.13 | Grubs | no - I just installed the patch just now and am about to test |
| 12:15.43 | morrece | iesh what is going on here |
| 12:16.15 | morrece | ok guys, I'm still stuck with this damn issue where I am trying to test my SIP softphone on my newly installed ASTERISK box |
| 12:16.22 | Mavvie | Grubs: thanks for pointing that out to me. |
| 12:16.58 | Peste | nobody has a clue how to fix this: asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_watchdog.so: undefined symbol: ast_load |
| 12:17.06 | morrece | I've created an 999 extention within extension.conf file that does an echo() which I can use to test connectivity |
| 12:17.12 | morrece | however I am getting nothing |
| 12:17.27 | Grubs | no probs. I just blew half the afternoon and a pile of frustration on it . |
| 12:17.57 | morrece | is this right, should my dialback at least call me back ... I am using X-lite |
| 12:18.12 | *** join/#asterisk SoloFlyer (n=soloflye@ppp146-82.lns3.adl2.internode.on.net) |
| 12:18.47 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 12:19.04 | SoloFlyer | t!seen |
| 12:19.08 | SoloFlyer | !seen |
| 12:19.36 | SoloFlyer | no bots? |
| 12:20.12 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:22.38 | *** join/#asterisk MadCamel (i=coder@janes.addicti0n.net) |
| 12:23.02 | coppice | ~seen |
| 12:23.22 | coppice | ~docs |
| 12:23.24 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:23.25 | AndyCap | SoloFlyer: /m nickserv info nick ? |
| 12:23.40 | coppice | hum. seen gave a private reply :-\ |
| 12:24.52 | MadCamel | I'm stumped. Anyone have * stop passing RTP packets as soon as the call is answered over an IAX trunk? ie: (IAX2/vpconnect-t02-4 is making progress passing it to SIP/101-84c0) <- RTP payload fine, first 1/2 second of voice on other end then (IAX2/vpconnect-t02-4 answered SIP/101-84c0)<- bang. RTP stops dead. no audio. |
| 12:25.10 | Mavvie | MadCamel: http://www.asterisk.org/node/55 |
| 12:25.49 | MadCamel | wtf kinda bug is that? haha |
| 12:25.51 | Grubs | I have a feeling that link is going to be posted alot! |
| 12:25.53 | MadCamel | thanks :) |
| 12:26.21 | Grubs | mine tests A-OK now....phew.! |
| 12:27.05 | ckruetze | Maybe somebody should change the title for this chat to include no audio bug and the link |
| 12:27.43 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 12:28.24 | MadCamel | Wouldn't be a bad idea, get some sort of publicity to the general user on the problem, I was awoken at 2am to "fix the goddamn phones" and have been tracing & debugging for many hours before finding this out. |
| 12:29.17 | morrece | please help me guys I'm batteling with this damn XLite Softphone |
| 12:29.28 | morrece | I cant seem to test the connectivity correctly |
| 12:29.42 | SoloFlyer | ~seen |
| 12:29.45 | morrece | I've configured sip.conf & extension.conf ... |
| 12:29.53 | morrece | what else should I be doing |
| 12:29.57 | oej | See you later, keep spreading the word about the patch! |
| 12:29.58 | morrece | must Xlite timesout |
| 12:30.08 | Grubs | oej - you rock! |
| 12:30.20 | *** join/#asterisk oceanlan (n=irc@cpe-71-65-112-38.woh.res.rr.com) |
| 12:31.10 | ravenpi | Anyone know how to reset the Grandstream-2000 to factory defaults? |
| 12:31.45 | cpm | so, 1.2.2 has sip-breakware in it? a sip timebomb, bizzare |
| 12:31.54 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 12:32.13 | cpm | "changing date of system to older time solves the problem." sheesh |
| 12:32.46 | MadCamel | yeah that's pretty wild |
| 12:33.36 | mutilator | wasn't just sip |
| 12:34.31 | Grubs | I guess here in AU we hit it first :p |
| 12:34.45 | cpm | mutilator, the patch says it's just sip calls |
| 12:35.07 | cpm | that is so very wierd. |
| 12:35.41 | Grubs | haha |
| 12:35.52 | cpm | I've noticed that about a lot of linux software, perhaps it's a linux bug? |
| 12:36.10 | tzanger | anyone else having really odd zaptel problems this morning? |
| 12:36.21 | sivana | morning |
| 12:36.22 | mutilator | probs with zap -> iax too |
| 12:36.28 | SoloFlyer | all of our phones had power (POE and switch is on ups) so i couldnt work out why our phones wouldnt work when we had a power outage |
| 12:36.41 | coppice | asterisk runs OK when I remove the power cord from my notebook. maybe they are a better option |
| 12:36.49 | sivana | heh |
| 12:36.56 | SoloFlyer | somehow the asterisk box managed to get pluged into a powerboard not connected to the ups |
| 12:37.01 | SoloFlyer | :( |
| 12:37.26 | *** part/#asterisk morrece (n=moreece@196.46.142.23) |
| 12:37.38 | coppice | most of the world power sockets are not connected to a UPS, so this is statistically highly likely |
| 12:37.53 | MadCamel | asterisk doesn't work so well when the UPS is on fire because some dumbass plugged a spaceheater into it either.. |
| 12:37.55 | SoloFlyer | should the bug be marked critical then ? |
| 12:38.18 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 12:38.20 | Grubs | no audio in a phone system is generally critical ! |
| 12:38.28 | backblue | hi, ppl |
| 12:38.33 | sivana | MadCamel: employee interviews and screening work best for that |
| 12:38.36 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 12:39.11 | MadCamel | sivana not that kinda company. it was my girl, actually. |
| 12:39.24 | sivana | hehe |
| 12:39.44 | *** part/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 12:39.44 | mutilator | didn't the ups say TILT when ya turned on the power for it? |
| 12:39.56 | mutilator | i know mine would annoy the hell outta ya with buzzes and beeps |
| 12:39.59 | sivana | ya, so she tilted it |
| 12:40.04 | SoloFlyer | lol |
| 12:40.09 | backblue | anyone with Ekiga and debian? |
| 12:40.17 | cpm | <PROTECTED> |
| 12:40.20 | *** join/#asterisk bofh42 (n=bofh42@p5482C14F.dip0.t-ipconnect.de) |
| 12:40.22 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 12:40.40 | mutilator | heh |
| 12:40.43 | MadCamel | it does that with minor overloads, but it's default action for major overload is to notify you bia visual, thermal, and olfactory means. |
| 12:40.43 | mutilator | yess |
| 12:40.55 | *** join/#asterisk JunK-Y (n=junky@67.71.108.7) |
| 12:41.02 | jozsab1 | how do i make my proxy to support 183 signal ? |
| 12:41.06 | mutilator | visual meaning smoking? |
| 12:41.16 | MadCamel | yep |
| 12:41.40 | AndyCap | Stupid beginner question: When I place a call through * from iaxcomm to visdn it will stuff up the progress info when I use u-law. (call is connected, but * still plays ringtone) but when I use gsm from iaxcomm -> * it works correctly. Anyone have a clue what to look for? |
| 12:42.25 | drumkilla | Critical Release: Asterisk 1.2.3, Coming Soon || Happy Birthday Cresl1n! |
| 12:42.31 | drumkilla | d'oh |
| 12:42.34 | *** topic/#asterisk by drumkilla -> Critical Release: Asterisk 1.2.3, Coming Soon || Happy Birthday Cresl1n! |
| 12:42.46 | mutilator | well |
| 12:42.52 | mutilator | it's about time for me to go home for the morning |
| 12:43.50 | sivana | drumkilla: what'd you do?! |
| 12:43.56 | zoa | THAT guy has a birthday again ??? |
| 12:43.59 | SoloFlyer | anyone know of anyone working on the ability to have asterisk test a route for latency and if it fails not route the call? |
| 12:44.45 | SoloFlyer | ie if call route >250ms fail to create sip connection, |
| 12:44.45 | MadCamel | SoloFlyer you could easily fudge that using qualify |
| 12:44.52 | wasim | zoa: cresl1n is from mercury |
| 12:45.13 | wasim | so its happy birthday every 88 days or so |
| 12:45.21 | zoa | that little bastard! |
| 12:45.28 | zoa | SoloFlyer: yes its called qualify = |
| 12:45.45 | SoloFlyer | hmm thanks guys |
| 12:45.48 | *** join/#asterisk pobre (n=seymore@203.215.73.192) |
| 12:45.53 | SoloFlyer | has that been in * long? |
| 12:46.36 | *** join/#asterisk pengyong (n=lala@218.93.159.101) |
| 12:47.05 | MadCamel | been there for at least 10 months, since I started using it |
| 12:47.05 | *** join/#asterisk zzxxcc (n=zzxxcc@221.235.45.176) |
| 12:48.19 | zzxxcc | does asterisk support voice modem? |
| 12:48.32 | coppice | not really |
| 12:48.40 | zzxxcc | i mean, the voice function of voice modem |
| 12:49.04 | coppice | voice modems don't do full duplex voice, so they are pretty useless |
| 12:49.57 | kink0 | I get on the gateway Layer1=ACTIVE and Layer2=Established for my E1 PRI port, but asterisk always fails reporting RED Alarm, what must I see now ? |
| 12:49.58 | zzxxcc | but asterisk does support analog telephony card? |
| 12:50.00 | SoloFlyer | MadCamel: 1.1.0.7? |
| 12:50.32 | coppice | zzxxcc: several analog telephony cards are supported |
| 12:51.09 | MadCamel | SoloFlyer beats me. your best bet is to look in the sample sip.conf and see if it has qualify= or not. voip-info has usage instructions |
| 12:51.50 | SoloFlyer | thanks didnt think of that... |
| 12:51.57 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-7-150.bredband.comhem.se) |
| 12:52.03 | cpm | the bug tracker claims 1.2.3 is out, I don't see it anywhere, clues? |
| 12:52.09 | SoloFlyer | the default sip.conf thing that is :) |
| 12:53.01 | zzxxcc | coppice:: the analog subscribe line signaling seems unreliable. how does the analog telephony card driver deal with this problem? |
| 12:53.12 | cpm | umm, i lied, sorry |
| 12:53.21 | coppice | zzxxcc: unreliable or incomplete? |
| 12:54.06 | backblue | anyone with zaphfc? |
| 12:54.14 | zzxxcc | coppice:: for example, an incoming call. the caller hang up, how could the voice modem know that? |
| 12:54.27 | bon | exten => _0.,2,SetCallerID(+1234${CALLERIDNUM}) |
| 12:54.30 | SoloFlyer | in case anyone other than me cares 1.1.0.7 has sip/iax qualify :) |
| 12:54.32 | bon | ^ should that work? :) |
| 12:54.38 | bon | the + seems to be problematic |
| 12:54.41 | SoloFlyer | thanks guys |
| 12:55.25 | coppice | zzxxcc: depends where you are. some places reverse the voltage. some create a timed break in the voltage. these days most just send a disconnect tone. the * tone detector kinda sucks, but detecting this tone can be very reliable |
| 12:56.02 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 12:56.08 | jozsab1 | how do i make my proxy to support 183 signal ? |
| 13:01.27 | *** join/#asterisk arkanis (n=test@80-219-10-141.dclient.hispeed.ch) |
| 13:03.06 | Grubs | cpm - its coming |
| 13:04.32 | backblue | there is anykind of problems using asterisk in a via epic motherboard? |
| 13:04.42 | *** join/#asterisk cricalixwood (n=rhill@hillconsult.force9.co.uk) |
| 13:05.20 | coppice | lots of people do. don't expect too much heavy compute stuff like echo cancellation or transcoding, though |
| 13:05.21 | zoa | backblue: no but you need to change the makefile |
| 13:06.21 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) || Happy Birthday Cresl1n! |
| 13:06.31 | arkanis | hi |
| 13:06.38 | arkanis | hope you can help me :) |
| 13:06.41 | coppice | why are those via boards and boxes so darned noisy? they seem to use the world's crappiest fans |
| 13:07.04 | backblue | drumkilla: where is that 1.2.3? |
| 13:07.11 | backblue | zoa: tks |
| 13:07.12 | arkanis | when I have "dial(SIP/166|20,r), what does the "20" mean? |
| 13:07.18 | drumkilla | backblue: the ftp |
| 13:07.31 | drumkilla | backblue: if you don't see it, give the mirrors a few more minutes to catch up |
| 13:07.48 | backblue | zoa: what happends if i dont change? because i'm using gentoo, and my wcfxo module, its allways crashing my machine, and i dont understand why. |
| 13:07.58 | jozsab1 | arkanis : i think the timeout until next action |
| 13:07.58 | sivana | arkanis: looks like timeout |
| 13:08.20 | backblue | drumkilla: no, tks, i will fetch it, but i allready compiled 1.2.2 with bristuff that allready have that patch. tks |
| 13:08.41 | Grubs | arkanis: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 13:08.43 | backblue | btw, i dont understand where is that error, i'm with 1.2.2 and i have audio normaly. |
| 13:08.48 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 13:08.50 | zoa | backblue: it just wont run |
| 13:09.02 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 13:09.08 | zoa | backblue: its locking up the pc ? |
| 13:09.13 | backblue | zoa: ho ok, so gentoo have changed the makefile then, tks |
| 13:09.30 | backblue | zoa: yes, a couple of hours i started wcfxo module, machine freezes. |
| 13:09.56 | arkanis | @grubs, I saw that, but I'm not sure ;-) |
| 13:10.02 | fugitivo | no audio? update your asterisk |
| 13:10.03 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:10.03 | fugitivo | lol |
| 13:10.07 | arkanis | thx @ others |
| 13:10.41 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 13:11.42 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:16.39 | *** join/#asterisk Grok_ (n=blah@c-71-196-75-163.hsd1.fl.comcast.net) |
| 13:17.27 | Grok_ | anyone here recommend a good telecom/pbx tutorial. |
| 13:17.51 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 13:17.58 | *** join/#asterisk pengyong (n=lala@222.185.18.133) |
| 13:19.14 | sm7xab | Grok:Just have an Asterisk HOWTO in 3 easy steps at http://www.automated.it/guidetoasterisk.htm |
| 13:19.23 | [TK]D-Fender | Grok_ : TFOT has a few interesting bits on it in the "foreward" section. |
| 13:19.27 | *** join/#asterisk lahaine (n=lahaine@116.68.119-80.rev.gaoland.net) |
| 13:19.33 | [TK]D-Fender | ~thebook |
| 13:19.34 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 13:20.53 | jozsab1 | how do i make my proxy to support 183 signal ? |
| 13:20.56 | Grok_ | thank you very much...greatly appreciated |
| 13:20.56 | Hmmhesays | so give you id card to the border guard |
| 13:21.10 | Grok_ | sorry for the delay was searching google as well |
| 13:21.13 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:23.04 | arkanis | Is it possible to have a php-application with phpagi on server1 and the asterisk-pbx on server2? |
| 13:23.39 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:25.40 | DarkFlibble | arkanis, fastagi supports that configuration iirc |
| 13:27.19 | arkanis | with "normal" phpagi not? it's curious because you can specify a server in phpagi-asmanger |
| 13:27.26 | arkanis | but I can't get it to work |
| 13:27.29 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:27.58 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 13:28.17 | DarkFlibble | arkanis, its been a while since I left asterisk use and am only gradulally getting back up to date with all the changes |
| 13:29.20 | DarkFlibble | fastagi was designed to use agi over a socket, so it should be usable on a second machine... the manager interface is another option for some tasks but when I left asterisk was fairly prone to crashes... |
| 13:29.47 | DarkFlibble | I am not familiar with any other external app methods |
| 13:30.40 | arkanis | hm, well so it could be that fastagi is rather slow |
| 13:30.40 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 13:30.40 | DarkFlibble | most external methods are slow... thye require spawning a new process... |
| 13:30.44 | festr_ | hello, anyone know, how to measure cpu usage (linux 2.6) on smp PC? :) vmstat 1 shows 41% idle. net-snmp and ucd_rawcpusystem shows another values. top show the same as vmstat 1. htop show cpu usage per cpu each 60% |
| 13:30.50 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 13:30.51 | festr_ | and load is 3 |
| 13:30.58 | festr_ | what does this mean at all |
| 13:31.03 | festr_ | how many cpu is free? :) |
| 13:31.17 | arkanis | but am I right, that phpagi is the best solution for a asterisk/php application? |
| 13:31.46 | DarkFlibble | load is the number of processes ready to run at any moment... so the higher it is the worse it is since processes are waiting to be scheduled |
| 13:32.08 | festr_ | DarkFlibble: so load about 3 on dual cpu mean what? |
| 13:32.35 | DarkFlibble | 3 processes are waiting to be scheduled to run... |
| 13:32.50 | festr_ | DarkFlibble: how is it possible if 40% of cpu remains? |
| 13:33.13 | fenlander | DarkFlibble: where are you giving your VoIP lectures? (saw your message earlier during the timebomb stuff) |
| 13:33.40 | DarkFlibble | since a modern multithreaded system is complex and not all resources can be served at all times... |
| 13:33.52 | DarkFlibble | fenlander, leicester, de montfort uni... |
| 13:34.07 | festr_ | DarkFlibble: could this change preemption? |
| 13:34.37 | festr_ | DarkFlibble: i've asterisk on dual xeon which handles about concurent 50-60 calls |
| 13:35.00 | fenlander | DarkFlibble: cool |
| 13:35.02 | DarkFlibble | festr_, you should run top and see which processes are running most of the time and work on these processes... |
| 13:35.20 | DarkFlibble | since that will result in best short term effects... |
| 13:35.40 | festr_ | DarkFlibble: so i'm asking my self, how much load i could accept and how to monitor if the system is overheaded |
| 13:35.48 | DarkFlibble | fenlander, not ran a lecture for years... so its gonna be strange... |
| 13:35.51 | festr_ | DarkFlibble: because load is 3 but vmstat 1 shows 40% idle |
| 13:36.10 | festr_ | DarkFlibble: most of time runs asterisk :) |
| 13:37.06 | festr_ | DarkFlibble: so i'm searching some docs how to measure system load, what values are critical and so on |
| 13:37.19 | DarkFlibble | festr_, have you got a real reason to try to reduce load or is it a percieved problem you are trying to fix? |
| 13:37.26 | fenlander | the system is overloaded when bad things start to happen... |
| 13:38.05 | fenlander | knowing how close you are to overlaod with a modern computer system is an ongoing subject of research |
| 13:38.09 | DarkFlibble | ie.. have you got bad call quality/ are sql requests taking tens of seconds etc... |
| 13:38.35 | Shaquile | [incoming] |
| 13:38.35 | Shaquile | exten => 3000,1,Macro(call_ext,200) |
| 13:38.52 | Shaquile | How can I get that incoming to forward the call to my cellphone? |
| 13:39.07 | DarkFlibble | Shaquile, use a pastebin if you are gonna paste a lot... |
| 13:39.24 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:39.33 | festr_ | DarkFlibble: i'm running net-snmp and monitor ucd_rawcpusystem which shows, that there are no cpu free in top a see, that asterisk is runnign 100% of cpu but vmstat 1 shows 40% idle. voice over asterisk is ok |
| 13:39.36 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 13:40.12 | festr_ | DarkFlibble: what values tells, there are problem with load? |
| 13:40.26 | Shaquile | DarkFlibble: Yes.. I didn't think that was too much tough.. |
| 13:40.28 | Shaquile | though |
| 13:40.41 | DarkFlibble | festr_, the main problem with high load on a real time system like asterisk is that calls quality suffers quickly |
| 13:40.57 | DarkFlibble | Shaquile, normally people paste 10 lines or so... |
| 13:40.59 | [TK]D-Fender | Shaquile : Just make it answer and dial out. Do you know how to Dial at all? |
| 13:41.11 | Shaquile | DarkFlibble: Ah okey =) |
| 13:41.49 | Shaquile | [TK]D-Fender: No not really, I was thinking of something like this.. |
| 13:41.50 | Shaquile | exten => 3000,1,Macro(call_out_digisip2,${PHONENUMBER},${CALLERIDNUM}) |
| 13:42.38 | festr_ | DarkFlibble: so how to measure cpu load so you can tell, is this point is limit |
| 13:42.47 | festr_ | s/is/in/ |
| 13:43.05 | arkanis | hm |
| 13:43.13 | festr_ | jbot: wow :) |
| 13:43.18 | jbot | it has been said that :) is honeypea |
| 13:43.18 | [TK]D-Fender | Shaquile : What is that macro? |
| 13:43.30 | Shaquile | [TK]D-Fender: Here's my extensions |
| 13:43.34 | Shaquile | [TK]D-Fender: http://pastebin.se/2203 |
| 13:44.22 | [TK]D-Fender | Shaquile : Make it look like one of your outgoing. |
| 13:44.44 | iCEBrkr | Yo yo |
| 13:45.28 | DarkFlibble | festr_, load on its own doesn't mean much... |
| 13:45.58 | *** part/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 13:46.02 | DarkFlibble | and there are many ways of measuring load that each try to show different things |
| 13:46.40 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 13:47.00 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 13:47.10 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 13:47.16 | [TK]D-Fender | Ma? |
| 13:47.23 | kink0 | I still getting RED alert on my Zap , even the E1 far end shows status=ACTIVE and level2=established |
| 13:47.35 | arkanis | When I have "ignorepat => 0" and I dial the number "0123456" it'ill cut of the 0 and gets "123456", is that right? |
| 13:47.56 | iCEBrkr | arkanis: That's how I've already read how it works. |
| 13:48.01 | ctooley | I must have missed something, what's the critical update that brought on 1.2.3? Is 1.2.1 affected? |
| 13:48.06 | iCEBrkr | arkanis: I can't say I've tried it tho. |
| 13:48.51 | iCEBrkr | 2006-01-25 09:46 +0000 [r8632] Olle Johansson <oej@edvina.net> |
| 13:48.52 | iCEBrkr | * channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster |
| 13:48.52 | iCEBrkr | <PROTECTED> |
| 13:48.55 | iCEBrkr | That doesn't sound good. |
| 13:48.57 | iCEBrkr | :) |
| 13:49.06 | synthetiq | auto fall through for channel "IAX2@addy/phone" Status is "UNKOWN" <--- what does this mean? |
| 13:49.17 | synthetiq | unknown |
| 13:49.26 | sivana | ctooley: has to do with brigding |
| 13:49.29 | synthetiq | im running * on fbsd |
| 13:49.34 | DarkFlibble | fenlander, still trying to work out everything I want to cover on the course tho... |
| 13:50.00 | kink0 | what signalling must I use to communicate with my 2N gateway ( E1 ) ? |
| 13:50.23 | kink0 | I have try pri_cpe and pri_net , but the same, still RED alert at Asterisk zap show status |
| 13:50.36 | tzanger | kink0: red alarm has nothing at all to do with signaling |
| 13:50.44 | iCEBrkr | kink0: You need to know how your lines is provisioned.. It should be simple from there |
| 13:50.44 | tzanger | that means asterisk is not physically seeing the other end |
| 13:50.53 | kink0 | tzanger, ahh ok, where must I see then ? |
| 13:50.54 | tzanger | use a loopback plug -- does the span go gren? |
| 13:50.58 | tzanger | er green |
| 13:51.03 | fenlander | DarkFlibble: depends on the audience i guess |
| 13:51.04 | iCEBrkr | GREN! |
| 13:51.12 | tzanger | does your gateway have the E1 interface up? |
| 13:51.22 | kink0 | iCEBrkr, the PRI is provisioned from a 2N Stargate gsm gateway, I set as TE/M |
| 13:51.29 | tzanger | what card are you using to communicate with the gateway? |
| 13:51.30 | ctooley | sivana, any idea if it's a bug that was in 1.2.1? |
| 13:51.37 | kink0 | tzanger, yes, is show as ACTIVE and OKESTB |
| 13:51.52 | kink0 | tzanger, TE405 from digium |
| 13:51.53 | tzanger | kink0: no idea what OKESTB means but ok. |
| 13:52.07 | kink0 | tzanger, is Layer2 = OK Established |
| 13:52.09 | tzanger | kink0: TE405 -- ok, did you move the jumper to put the span in E1 mode (as opposed to T1) ? |
| 13:52.11 | sivana | ctooley: I'm not sure.. I think it's just 1.2.2 |
| 13:52.43 | kink0 | tzanger, opsss... I did not... I have not touch hardware at the TE405 card |
| 13:52.59 | tzanger | kink0: that's likely the first issue. :-) |
| 13:52.59 | DarkFlibble | fenlander, thats what I was thinking... thought about writing a syllibus that could cover 3 days and dropping stuff depending on levels of the attendees since its a little of a moving target atm |
| 13:53.31 | tzanger | DarkFlibble: nice. kind of a "choose your own" training course :-) |
| 13:53.36 | kink0 | tzanger, right !!! sometimes we start at the most complicate scenary when may be the most simple... ok, will check it now. |
| 13:54.08 | iCEBrkr | :D |
| 13:55.31 | Katty | hmm. |
| 13:55.52 | tzanger | fenlander: that is torture |
| 13:56.10 | fenlander | tzanger: couldn't agree more |
| 13:56.11 | tzanger | but very, very good training |
| 13:56.11 | coppice | fenlander I keep wanting to make funny faces to the camera when they do that :-) |
| 13:56.12 | DarkFlibble | yeah... I wouldn't choose to be videoed... |
| 13:56.13 | tzanger | trial by fire |
| 13:56.39 | tzanger | Katty: come on baby light my fire |
| 13:56.46 | Katty | tzanger: welcome to mount wannahockaloogie! |
| 13:56.51 | tzanger | Katty: hahahaha |
| 13:56.53 | tzanger | I love that movie |
| 13:56.55 | coppice | tzanger: hey, you beat me to it |
| 13:57.05 | tzanger | "hey you speak whale? I wish I could do that!" |
| 13:57.19 | tzanger | coppice: score one for me :-) |
| 13:57.31 | [TK]D-Fender | Katty: Mew. |
| 13:57.40 | Katty | [TK]D-Fender: mew. |
| 13:57.46 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 13:58.55 | [TK]D-Fender | iCEBrkr : Got a look at Jim's "inherited" setp.... I want the crack its designer was on..... |
| 13:59.02 | Paulo | somebody knows if there are any variables I can pickup from rx_fax with information about the fax? (number of pages, etc) |
| 13:59.06 | tzanger | [TK]D-Fender: what's that |
| 13:59.09 | iCEBrkr | [TK]D-Fender: LOL |
| 13:59.16 | tzanger | Paulo: show application rxfax |
| 13:59.19 | iCEBrkr | [TK]D-Fender: Jim always ends up with jobs/projects like that. |
| 13:59.29 | Paulo | tzanger: thanks. |
| 14:00.15 | iCEBrkr | [TK]D-Fender: He's got that grey cloud over him at all times. Bad luck seems to follow him everwhere, due just can't get a break |
| 14:01.46 | [TK]D-Fender | iCEBrkr : Well lets say its not unrecoverable, but in need of a "many pass" cleanup given its size. I could shink the setup by half in about an hour or so. |
| 14:02.07 | iCEBrkr | [TK]D-Fender: Cool stuff |
| 14:02.26 | *** join/#asterisk oceanlan (n=irc@cpe-71-65-112-38.woh.res.rr.com) |
| 14:04.06 | arkanis | I don't really understand for what ignorepat is useful |
| 14:04.51 | arkanis | in the asterisk documentation they have an example with ignorepat, but it's rather useless, because the strip the first digit of with ${EXTEN:1} |
| 14:05.04 | oceanlan | anyone know if there is an AMP irc? |
| 14:05.15 | arkanis | can somebody tell me for what that ignorepat is really usefull |
| 14:05.18 | iCEBrkr | oceanlan: No, but I wish it was easier to find :P |
| 14:05.52 | DarkFlibble | arkanis, it doesn't lose the dialtone when you start dialling iirc |
| 14:06.00 | [TK]D-Fender | oceanlan : #amportal |
| 14:06.12 | DarkFlibble | ie... if you dial 9 for an outside line you still have the dialtone... |
| 14:06.42 | tzanger | arkanis: I think that ignorepat just configures what the PBX dialtone ignores |
| 14:06.47 | arkanis | a well |
| 14:06.52 | tzanger | i.e. ignorepat=9 means that if you pick up and dial 9 you still hear dialtone |
| 14:07.01 | [TK]D-Fender | arkanis : Thats only useful for ZAP analog phones. Thats so if your setup has the caller dial "9" before the number to dial out for instance, that it will continue to provide dial-tone after the 9 is received, but not after the next digit. |
| 14:07.05 | tzanger | then dial 4 (for example) and it stops dialtone |
| 14:07.15 | tzanger | see I am smart :-) |
| 14:07.18 | tzanger | s m r t |
| 14:07.38 | Katty | tzanger: you so are. |
| 14:07.57 | iCEBrkr | So.. I started an Asterisk PBX group on MeetUp.com |
| 14:08.06 | iCEBrkr | Hrrm, now to promote the thing. |
| 14:08.06 | tzanger | iCEBrkr: heh |
| 14:08.11 | tzanger | we did that in toronto |
| 14:08.19 | tzanger | then they started charging so we said fuck it |
| 14:08.31 | iCEBrkr | tzanger: Well, I've been trying to organize a Asterisk Users Group. |
| 14:08.43 | iCEBrkr | tzanger: Yeah, what's with that? I didn't realize you had to pay for Meetup.com |
| 14:08.53 | iCEBrkr | Oh well. No biggie |
| 14:09.05 | iCEBrkr | My boss said if it's an issue he'll take over the payments. |
| 14:10.12 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 14:10.15 | DarkFlibble | how many people in here besides me and fenlander are in the UK? just curious... |
| 14:10.18 | *** join/#asterisk MrChimpy (n=MrChimpy@smtp-gw.amplefuture.com) |
| 14:10.20 | Modcuts | is there a way i can check that the g729 is install properly? as i only seem to have one sided calls when i use it as the codec of choice, any ideas? |
| 14:10.47 | kshumard | Modcuts, `show g729` on the asterisk cli |
| 14:11.11 | MrChimpy | paulo: bom dia! |
| 14:11.22 | iCEBrkr | Getting g729 isn't all that great when VoicePulse and other providers don't support it. |
| 14:11.36 | fenlander | DarkFlibble: not many then :) |
| 14:11.43 | lancey | ? |
| 14:11.45 | Modcuts | yep that gives me the "0/0 encoders/decoders of 5 licensed channels currently in use" |
| 14:11.46 | DarkFlibble | fenlander, at least not live ones... |
| 14:11.52 | lancey | iCEBrkr: they don't? |
| 14:11.54 | iCEBrkr | Modcuts: Then it's installed. |
| 14:11.59 | iCEBrkr | lancey: Nope. |
| 14:12.08 | lancey | strange |
| 14:12.19 | MrChimpy | i'm getting regular -- B-channel 0/1 successfully restarted on span 1 etc etc on a TE410... is that bad? it looks it... it apparently works though. |
| 14:12.23 | Modcuts | um..... could one sided voice be down to nat when all other codecs work fine? |
| 14:12.42 | iCEBrkr | lancey: Asterisk will spew the g729 "out of licenses" message when on a call and you'll get one-way audio |
| 14:12.49 | robin_sz | OK, * keeps quitting on me ... |
| 14:12.51 | iCEBrkr | Modcuts: Sure. |
| 14:13.04 | iCEBrkr | robin_sz: It's tired and wants to take a nap |
| 14:13.14 | iCEBrkr | robin_sz: Or you're not paying it enough.. I'd quit too. |
| 14:13.28 | kshumard | Modcuts, what * version? Which g729 codec do you have installed? I mean, which architecture? |
| 14:13.39 | robin_sz | the only clue is a CAPI illegal controller in an aoutgoing call attempt a bit earlier |
| 14:13.57 | iCEBrkr | robin_sz: Run 'asterisk -cvvvvvvvv' |
| 14:14.07 | lancey | iCEBrkr ? |
| 14:14.12 | Modcuts | 32bit athlon xp |
| 14:14.50 | iCEBrkr | lancey: When you make a call via Voicepulse and use g729, you'll only get one-way audio and the CLI is spammed with a message stating you're out of licenses or something |
| 14:14.50 | *** join/#asterisk HamYaI (i=HamYai@125.24.12.180) |
| 14:15.01 | iCEBrkr | lancey: Since you need the codec at both ends of the call |
| 14:15.05 | Paulo | How can I list all variables in a given context? |
| 14:15.16 | iCEBrkr | Paulo: WTF?! |
| 14:15.17 | Modcuts | kshumard : and my cpu iis a AMD Sempron(tm) 2600 |
| 14:15.24 | kshumard | Modcuts, what * version? |
| 14:15.29 | robin_sz | iCEBrkr: running with -rvvvvvvv already ... |
| 14:15.30 | lancey | iCEBrkr i |
| 14:15.30 | HamYaI | ohh, 1.2.3 is released glad to hear that! |
| 14:15.40 | Paulo | iCEBrkr: for debuging purposes |
| 14:15.41 | iCEBrkr | robin_sz: Alrighty. |
| 14:15.45 | lancey | iCEBrkr i haven't asked anything about g729 |
| 14:15.55 | lancey | except wondering that voicepulse doesn't support it |
| 14:15.56 | iCEBrkr | Paulo: You should be aware of most variables in your dialplan. |
| 14:16.10 | iCEBrkr | Paulo: There's no way to dump all variables... You'd have to jam in a bunch of NoOp() to see them. |
| 14:16.29 | iCEBrkr | lancey: Well, if you think about all the calls they handle, they'd need a license for each call. |
| 14:16.39 | Paulo | iCEBrkr: I need a NoOp(${FOO}) for each variable? |
| 14:16.50 | iCEBrkr | Paulo: Yea |
| 14:16.52 | tzanger | iCEBrkr: it's not expensive I don't think but anyway |
| 14:17.07 | HamYaI | anyone running astguiclient on FC3 at all? |
| 14:17.08 | lancey | iCEBrkr: $10 one-time musn't be an issue for voicepulse |
| 14:17.11 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 14:17.12 | Math[laptop] | iCEBrkr: show channel [chan] dumps them |
| 14:17.15 | iCEBrkr | tzanger: $10 a pop? x 1000 channels? |
| 14:17.23 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 14:17.27 | tzanger | iCEBrkr: ? for meetup.com? |
| 14:17.27 | iCEBrkr | Math[laptop]: Good to know! |
| 14:17.29 | lancey | well, they are not a charity |
| 14:17.32 | Modcuts | kshumard : verson 1.2.0 |
| 14:17.37 | oceanlan | I have read that I should turn SELinux off...is this in /ect/grub.conf? |
| 14:17.37 | Paulo | iCEBrkr: thanks. I'm very newbie. |
| 14:17.41 | iCEBrkr | tzanger: Oh, shit, we got topics crossed :P |
| 14:17.48 | *** join/#asterisk Cyon (n=cyon@cyons.net) |
| 14:17.52 | tzanger | iCEBrkr: hahaha |
| 14:18.15 | iCEBrkr | tzanger: meetup.com is $20/mo if you go month-by-month, but if you pay for the year, it's really not that bad. |
| 14:18.16 | [TK]D-Fender | "But Egon.. I though you said crossing the streams was bbbaaaaddddd..." |
| 14:18.22 | iCEBrkr | [TK]D-Fender: hehe |
| 14:18.26 | kshumard | Modcuts, would you be able to try 1.2.1 temporarily to see if it works there? I've heard some people say that g.729 breaks between 1.2.1 and 1.2.2 but I haven't been convinced yet |
| 14:18.27 | *** part/#asterisk SoloFlyer (n=soloflye@ppp146-82.lns3.adl2.internode.on.net) |
| 14:18.52 | Modcuts | that would envolve me doing a complete reinstall? |
| 14:18.53 | oceanlan | how do I turn OFF SELinux? |
| 14:19.19 | Katty | use the power button. |
| 14:19.26 | Math[laptop] | thats a way |
| 14:19.48 | [TK]D-Fender | oceanlan : Try asing in #linus or #centos or something..... |
| 14:19.51 | oceanlan | hah, funny...i dont see any entry in grub loader...should I add my owm selinux=off? |
| 14:19.53 | [TK]D-Fender | #linux |
| 14:20.03 | oceanlan | thanks! |
| 14:20.11 | [TK]D-Fender | oceanlan : Maybe try http://www.google.com |
| 14:20.24 | Peste | what can i do if: [chan_zap.so]Jan 25 16:19:43 WARNING[27486]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart |
| 14:20.25 | Peste | Jan 25 16:19:43 WARNING[27486]: loader.c:554 load_modules: Loading module chan_zap.so failed! |
| 14:20.42 | Math[laptop] | Peste: check libpri version |
| 14:20.52 | oceanlan | I did google..got lots of different results..all of them say its in Grub, but none of them say what to do if it is not there |
| 14:21.26 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 14:21.48 | *** join/#asterisk Weezey (n=chatzill@tiny.tsnetworks.ca) |
| 14:22.06 | Modcuts | kshumard : can i just update asterisk or just that remove all the configs? |
| 14:22.24 | Weezey | I'm having a no-way audio problem. |
| 14:22.33 | Math[laptop] | Weezey: update to 1.2.3 |
| 14:22.35 | iDunno | get someone to speak, then ;) |
| 14:22.36 | [TK]D-Fender | No way! |
| 14:22.37 | Katty | ata-way problem! |
| 14:22.45 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:22.46 | Weezey | math: shiza |
| 14:22.48 | kshumard | Modcuts, best bet would be to checkout 1.2.1 into a new dir, while leaving your current asterisk build intact |
| 14:22.56 | kshumard | Modcuts, configs won't be overwritten |
| 14:23.06 | Peste | Math[laptop]: how do i check it? changelog shows vers. 0.3.0 |
| 14:23.15 | Katty | asteriskmonkey: morning. |
| 14:23.25 | asteriskmonkey | can anyone help me debug why my span dsp wont load? i compiled it with no errors so i am lost to the next step in the debug process. |
| 14:23.30 | Katty | [TK]D-Fender: is show version /supposed/ to give me a number? |
| 14:23.51 | Math[laptop] | Weezey: there's a bug in the audio bridging in 1.2.x that happens in an overflow in date functions... and today's the date when its overflowing |
| 14:24.34 | Hmmhesays | heh |
| 14:24.40 | Weezey | math: oie vey. |
| 14:24.50 | Weezey | time yesterday |
| 14:24.50 | Katty | Hmmhesays: ! |
| 14:24.56 | Hmmhesays | Katty |
| 14:24.57 | Katty | Hmmhesays: also, hi. |
| 14:24.57 | Hmmhesays | how are you? |
| 14:25.03 | Katty | just peachy keen. |
| 14:25.07 | Hmmhesays | good good |
| 14:25.08 | *** join/#asterisk dwildes (n=dwildes@209.164.237.195) |
| 14:25.09 | Katty | well, no i take that back. |
| 14:25.15 | Katty | wet seal did not have the shirt in i wanted last night. |
| 14:25.41 | Katty | but other than that, i'm just peachy keen. |
| 14:25.54 | Hmmhesays | awww thats to bad |
| 14:25.58 | Hmmhesays | i don't buy clothes anymore |
| 14:26.03 | Katty | Hmmhesays: isn't it great? i'm having girly outbursts. |
| 14:26.12 | Hmmhesays | i just keep finding old stuff in my closet... |
| 14:26.18 | Hmmhesays | Katty: i dunno its a bit disturbing |
| 14:26.19 | Katty | Hmmhesays: maybe someday i'll look girly too. |
| 14:26.24 | [TK]D-Fender | Katty : It tends to say what SVN release and compile date I believe |
| 14:26.24 | Katty | Hmmhesays: i know. |
| 14:26.35 | Hmmhesays | My wireless monitors are on the way |
| 14:26.42 | Hmmhesays | probably won't get here till monday though |
| 14:26.58 | Drew | what are you doing with wireless monitors? wardriving? |
| 14:27.03 | Katty | [TK]D-Fender: Asterisk CVS-HEAD built by root@BES.COPI-RITEINC on a i686 running Linux on 2005-06-08 18:35:10 <- that doesn't appear to have a release number on it. )= |
| 14:27.12 | Hmmhesays | Drew: wireless ear monitors for the band |
| 14:27.17 | Paulo | I have an E1 with 30 lines and 500 numbers from a Brazilian telco. I would like to know the dialed number, but DNID is empty. |
| 14:27.20 | Drew | ah ^^ |
| 14:27.25 | Hmmhesays | replacing the shiatty floor wedges |
| 14:27.42 | Hmmhesays | free's up an amp too |
| 14:27.51 | Katty | i'll take the amp. |
| 14:28.02 | Katty | could use a new one for the strat |
| 14:28.11 | iCEBrkr | Geesh my boss is paranoid. I've just installed 3 safety nets to prevent any calls from leaving the building between a set time period... |
| 14:28.17 | Hmmhesays | to bad its a poweramp for PA, not a guitar amp |
| 14:28.23 | Peste | Math: i have the newest vers of libpri from digium cvs |
| 14:28.39 | asteriskmonkey | span dsp dont know why it wont load anyone got any ideas where to look? everything compiled ok. |
| 14:29.04 | *** join/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74) |
| 14:29.24 | FlipZZZ | hello all |
| 14:29.32 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 14:29.39 | [TK]D-Fender | Katty : Oh well.. not sure what to say.... check your source folder then. |
| 14:29.59 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 14:31.18 | [TK]D-Fender | Hmmhesays : A benifit for my using a GT-8... full amp / speaker simulator and I can run it through a driect box just fine :) |
| 14:31.25 | *** join/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 14:31.41 | tzafrir_laptop | hmmm, for those of you not busy upgrading, try 'help add extension' |
| 14:31.55 | tzafrir_laptop | The example there is wrong, right? |
| 14:32.15 | Hmmhesays | [TK]D-Fender yeah i'm going to pick up a podxt live for the same purpose I think, but that doesn't do any good live |
| 14:32.34 | Cyon | tzafrir_home: No idea; never used it. :-P |
| 14:32.53 | tzafrir_laptop | That is: It should be 'IAX2' and not 'IAX' |
| 14:33.09 | jozsab1 | how do i make my proxy to support 183 signal ? |
| 14:33.11 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 14:33.13 | tzafrir_laptop | Not that I have any intention of trying it |
| 14:33.23 | Cyon | tzafrir_home: Yep, you are right |
| 14:33.34 | [TK]D-Fender | Hmmhesays : What do you mean "no good"? Pod's are pretty decent. Mind you Behringer is a much better bang/buck. |
| 14:34.10 | iCEBrkr | Damnit! There's no mention of app_amd anywhere |
| 14:34.32 | Weezey | the perils of living on the bleeding edge. |
| 14:34.57 | tzanger | :-) |
| 14:35.21 | Katty | sounds kinky. |
| 14:35.30 | iCEBrkr | Katty: Pervert |
| 14:35.39 | Katty | [TK]D-Fender: glint? |
| 14:35.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:35.45 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
| 14:35.53 | iCEBrkr | GNU lint |
| 14:36.06 | jlewis | anyone aware of a bug in 1.2.2 that might cause a customer with both sip and zap (PRI to PBX) phones to be able to place calls (IAX2) over our network, but get no audio in either direction after the call connects? |
| 14:36.07 | [TK]D-Fender | Katty: http://dictionary.reference.com/search?q=glint #1 |
| 14:36.17 | donnib | hi all. i have a simple qiestion regarding two different SIP providers. Can somebody help me ? |
| 14:36.21 | jlewis | downgrading them to 1.2.0 got rid of it |
| 14:36.35 | Katty | [TK]D-Fender: oh. |
| 14:36.41 | [TK]D-Fender | jlewis : How about upgrading to 1.2.3 then? |
| 14:36.46 | Katty | [TK]D-Fender: goshdangitanyhow. use bigger words. |
| 14:36.47 | jlewis | and what's odder, is this system was working fine yesterday under 1.2.2 |
| 14:37.05 | Math[laptop] | jlewis: we're aware of that bug in 1.2.2, but it has been fixed in 1.2.3 |
| 14:37.08 | [TK]D-Fender | Katty : I don't want you in convulsions now do I? ..... ;) |
| 14:37.13 | jlewis | didn't realize it was out...and I've been running 1.2.0 myself for a while, so I had some confidence in it |
| 14:37.17 | Assid | 1.2.3 out? |
| 14:37.20 | Math[laptop] | jlewis: see http://bugs.digium.com/view.php?id=6349 (this is also the reason of the fast release of 1.2.3) |
| 14:37.26 | [TK]D-Fender | TOPIC DAMNIT! |
| 14:37.28 | Katty | [TK]D-Fender: uhm. |
| 14:37.31 | Assid | holy crap |
| 14:37.32 | Math[laptop] | [09:37] * Topic is 'Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) || Happy Birthday Cresl1n!' |
| 14:37.33 | donnib | my question is: the two providers i have runs on two different ports 5060 and 1737. How do i tell Asterisk to use port 1737 for one of the providers ? |
| 14:37.35 | Katty | [TK]D-Fender: err. |
| 14:37.38 | Katty | [TK]D-Fender: sure? |
| 14:37.48 | [TK]D-Fender | Katty: rawr :D |
| 14:37.56 | iCEBrkr | Are there any plans on implementing VoiceXML into Asterisk? Anyone know? |
| 14:38.11 | [TK]D-Fender | iCEBrkr : Let them finish implementing SIP first please! |
| 14:38.14 | brad_mssw | crap, 1.2.3 now |
| 14:38.17 | iCEBrkr | [TK]D-Fender: hehe |
| 14:38.20 | Katty | [TK]D-Fender: kthx. |
| 14:38.28 | Math[laptop] | 1.2.1 aint affected, only 1.2.2 is |
| 14:38.37 | jlewis | wow...no more audio after a certain date?!? |
| 14:38.40 | iCEBrkr | cpm: Oh come'on! XML is your friend! |
| 14:38.46 | Math[laptop] | jlewis: overflow in date functions heh |
| 14:38.49 | iCEBrkr | :) |
| 14:38.55 | Math[laptop] | jlewis: today was the magic date |
| 14:38.56 | donnib | anyone ? |
| 14:39.00 | *** join/#asterisk SrFr (n=blaat@cust.12.229.adsl.cistron.nl) |
| 14:39.10 | jlewis | that's too funny to be true |
| 14:39.11 | Math[laptop] | http://lists.digium.com/pipermail/asterisk-dev/2006-January/018245.html |
| 14:39.16 | iCEBrkr | donnib: That sounds kinda odd to me that your providers aren't using the standard SIP/IAX ports |
| 14:39.48 | jlewis | now = 1138199964...how'd that overflow? |
| 14:40.03 | jlewis | looking at the patch now |
| 14:40.35 | *** join/#asterisk I-MOD (n=opticron@pcp01931374pcs.huntsv01.al.comcast.net) |
| 14:40.53 | Katty | make the phones stop ringing :< |
| 14:40.57 | donnib | iCEBrkr: well that how it is. it works fine from my sipura on port 1737 |
| 14:41.14 | kshumard | Katty, well, we made the audio stop after you pick up the phones... is that close enough? ; ) |
| 14:41.27 | jlewis | guess that bug was introduced after 1.2.0 |
| 14:41.35 | Math[laptop] | jlewis: introduced in 1.2.2 |
| 14:41.39 | Math[laptop] | even 1.2.1 is fine |
| 14:41.57 | donnib | iCEBrkr: on configuration page it says : SIP Port: 1737 |
| 14:42.30 | donnib | iCEBrkr: have any idea how to tell Asterisk to run that sepcifically provider on that port ? |
| 14:43.01 | Katty | kshumard: no |
| 14:43.10 | Katty | kshumard: that simply Will Not Do |
| 14:43.15 | iCEBrkr | donnib: I'd instinctively put :1737 at the end of the IP address of your provider.. But I'm not quite sure Asterisk works that way |
| 14:44.48 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:44.50 | jlewis | ok...will download 1.2.3 now and start compiling |
| 14:44.58 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 14:45.10 | donnib | iCEBrkr: will try that. thanx |
| 14:47.59 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 14:48.19 | LostFrog | What would cause static on a SIP->SIP call? |
| 14:48.41 | iCEBrkr | Shitty phones :) |
| 14:49.01 | *** join/#asterisk techy (n=tecky@66.9.96.115) |
| 14:49.01 | LostFrog | This just started happening |
| 14:49.46 | [TK]D-Fender | LostFrog: across how many different phones? |
| 14:49.57 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:50.11 | LostFrog | One phone to 3 other phones. |
| 14:50.28 | LostFrog | Also incoming calls from analog to the same 3 phones. |
| 14:52.12 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 14:54.30 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 14:54.37 | LostFrog | It seems to be only the phones that are out of the country. :( |
| 14:54.53 | arkanis | in my extensions.conf under [globals] i have defined "TIMEOUT => 20", when I call exten => 1234,1,Dial(SIP/121|${TIMEOUT},th) it doesnt work, does somebody know why? |
| 14:55.06 | MrChimpy | lf: isp link problems? |
| 14:55.29 | LostFrog | Why would that cause static, wouldn't that cause distortion and dropped packets? |
| 14:56.10 | *** join/#asterisk tomben (n=tomben@fw01.ext.atl.jboss.com) |
| 14:56.18 | MrChimpy | only a suggestion. could be codec getting confused |
| 14:56.24 | arkanis | ah, sorry meant [general] |
| 14:56.39 | LostFrog | Hmm.. let me check the codec. |
| 14:56.40 | Peste | i got a new one :( [chan_capi.so]Jan 25 16:53:49 WARNING[7068]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_pthread_create |
| 14:56.40 | Peste | Jan 25 16:53:49 WARNING[7068]: loader.c:554 load_modules: Loading module chan_capi.so failed! |
| 14:56.40 | corruptor | arkanis, you need TIMEOUT=20 |
| 14:56.45 | *** part/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 14:58.47 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 15:01.18 | tzanger | I am too stupid to live |
| 15:01.28 | tzanger | sangoma's wanpipe utilities weren't installing |
| 15:01.35 | tzanger | ... I didn't install the C++ compiler |
| 15:01.48 | tzanger | <-- idiot |
| 15:01.49 | Math[laptop] | lol :P |
| 15:01.49 | LostFrog | ulaw -> ulaw doesn't really improve the static. |
| 15:02.03 | tzanger | the worst of it is that I had sangoma on the phone when I discovered that |
| 15:02.30 | Math[laptop] | I just called the boss of my phone provider to tell him "Hi! Upgrade your asterisk server please!" |
| 15:02.34 | pigpen | tzanger, you need a beer...or coffee (whichever is your vice) |
| 15:02.35 | Math[laptop] | the guy's like "who the fuck are you?" |
| 15:02.40 | corruptor | Peste, i think you don't need chan_capi. Disable it in modules.conf |
| 15:02.54 | tzanger | pigpen: it's 10 in the morning... beer sounds good. :-) |
| 15:03.03 | tzanger | Math[laptop]: hahaha |
| 15:03.22 | tzanger | I love sangoma's S518 PCI ADSL card |
| 15:03.24 | LostFrog | Beer always sounds good. |
| 15:03.36 | MrChimpy | are the sangoma utils any good with the digium card? |
| 15:03.43 | Math[laptop] | alcohol and morning don't sound good to me |
| 15:03.58 | tzanger | MrChimpy: I don't use the T1/E1 products from sangoma... I have an A102u and it works, but I find it no better or worse than the digium kit |
| 15:03.58 | pigpen | tzanger, 9:00am here....and I am already wanting to break into the liquor cabinet...and I don't drink. |
| 15:04.19 | tzanger | hahaha |
| 15:04.20 | [TK]D-Fender | tzanger : I love my S518 too :) much cleaner server thanks to it... |
| 15:04.32 | tzanger | [TK]D-Fender: amen to that, and QoS that works amazingly well |
| 15:04.36 | [TK]D-Fender | Katty: no use... I sleep the sleep of the righteous :D |
| 15:04.40 | tzanger | at least until I saturate the uplink with tagged traffic :-) |
| 15:04.49 | NewSole | so did we happen to fix the timebomb in asterisk 1.2.2 |
| 15:05.03 | tzanger | "What do you mean QoS won't give us more than 800kbps upstream?" |
| 15:05.04 | zamsler | lol @ saturate. |
| 15:05.08 | tzanger | NewSole: yes it was fixed VERY quickly |
| 15:05.16 | zoa | NewSole: yes |
| 15:05.19 | NewSole | what was it |
| 15:05.19 | zoa | download 1.2.3 |
| 15:05.26 | tzanger | NewSole: ariane 5 rocket problem |
| 15:05.30 | zoa | http://lists.digium.com/pipermail/asterisk-dev/2006-January/018245.html |
| 15:05.35 | tzanger | exact same scenario ... out of bounds condition not checked |
| 15:05.47 | [TK]D-Fender | MrChimpy : Compare the specs, ask people who've used... all the ususals.... |
| 15:06.23 | MrChimpy | aye. we didn't know of sangoma before getting the TE411 |
| 15:06.31 | ErMeS|Work | hi all |
| 15:06.34 | Katty | [TK]D-Fender: no fun :< |
| 15:06.45 | ErMeS|Work | is there any calls register for asterisk ? |
| 15:07.32 | ErMeS|Work | something which does a calls list |
| 15:07.34 | [TK]D-Fender | Katty: The GF is jealous of that. No nightmares, and I think only once or twice a year have nights wher I just can't sleep. |
| 15:07.48 | NewSole | well dont blame me if I dont jump on band wagon and upgrade..... after I did upgrade to 1.2.2 all our production servers went dead.... |
| 15:08.10 | NewSole | i have 1.2.0 working well... now.... |
| 15:08.26 | tzanger | 1.2.0 should be fine |
| 15:08.53 | LostFrog | I'm still running rc2 |
| 15:09.13 | Math[laptop] | *only* 1.2.2 is affected |
| 15:09.23 | *** join/#asterisk moy (n=kvirc@dsl-201-133-179-254.prod-infinitum.com.mx) |
| 15:09.37 | NewSole | between digiums con of buying echo cancel TE cards and now this.... digium lost all our trust |
| 15:09.52 | tzanger | NewSole: ... which con is that, precisely? |
| 15:09.55 | fourcheeze | how do I stop people using our asterisk server as an outbound proxy while still allowing incoming calls from anonymous sip clients? |
| 15:10.05 | MrChimpy | what's wrong with the echo cancelling cards? |
| 15:10.06 | tzanger | fourcheeze: sounds like some context magic |
| 15:10.18 | MrChimpy | they're a heck of a lot cheaper than what we used to use |
| 15:10.30 | fourcheeze | tzanger: I think it must be but I can't think what |
| 15:10.35 | Katty | [TK]D-Fender: lucky you. |
| 15:10.41 | NewSole | to spend the extra 1000$ to upgrade our current TE410 cars ro TE411P cards |
| 15:10.46 | *** part/#asterisk zzxxcc (n=zzxxcc@221.235.45.176) |
| 15:10.47 | Katty | [TK]D-Fender: i've been waking up ever couple hours here of late. |
| 15:10.52 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 15:10.53 | tzanger | fourcheeze: well the context that you dump incoming SIP people to should not have any way of dialing out |
| 15:10.57 | tzanger | fourcheeze: pretty straightforward |
| 15:11.00 | tzanger | i.e. |
| 15:11.02 | tzanger | [sip-incoming] |
| 15:11.04 | NewSole | and they made no diff acualy echo was worse |
| 15:11.06 | fourcheeze | tzanger: no it's not dialling out |
| 15:11.07 | tzanger | include = extensions |
| 15:11.13 | MrChimpy | newsole: so return them? |
| 15:11.15 | tzanger | fourcheeze: ok I don't understand the problem then |
| 15:11.17 | fourcheeze | it's using it as a proxy |
| 15:11.22 | *** join/#asterisk moy (n=kvirc@dsl-201-133-179-254.prod-infinitum.com.mx) |
| 15:11.22 | tzanger | NewSole: how is that a con, exactly? |
| 15:11.37 | hackeron | hey, my asterisk box ran out of diskspace which killed asterisk - I made some but now asterisk just takes 99.9% cpu and while it answers calls, there's no audio. Any ideas? |
| 15:11.54 | fourcheeze | tzanger: e.g. I have a sip client that's not registered anywhere, and I put our asterisk server as a sip proxy |
| 15:12.03 | NewSole | cant return them... boxes will not have timing source then cause they took our old TE410 cards |
| 15:12.03 | fourcheeze | and then give my sip client a sip url |
| 15:12.40 | [TK]D-Fender | NewSole : ZTDUMMY |
| 15:12.49 | NewSole | so now all they are doing in our box now is just supplying Timing..... |
| 15:13.04 | NewSole | ZTDummy is only good for 20 calls tops |
| 15:13.05 | tzanger | NewSole: how can you not return them? Just RMA them for four TE410 + difference in price... honestly |
| 15:13.16 | [TK]D-Fender | NewSole : You mean your TE411P are only doing timing at this point? |
| 15:13.35 | NewSole | ya we bought 12 Vega Stream 400's |
| 15:13.36 | tzanger | NewSole: ?? how the hell is there a number of calls limitation with ztdummy? |
| 15:13.57 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:14.10 | [TK]D-Fender | NewSole : So the Vegas are PRI -> SIP? |
| 15:14.11 | tzanger | NewSole: secondly you can disa ble the VPM with vpmsupport=no or phyusically pull the VPMs off the TE411s and hey, presto! TE410s! ship the VPMs back to digium for refund |
| 15:14.27 | tzanger | NewSole: it sounds like you *want* to have a beef with Digium rather than get the problem solved |
| 15:14.52 | asteriskmonkey | anyone have any idea why sip calls whould sunddely only start doing 1 way audio streams |
| 15:15.07 | NewSole | no at this point we can not afford down time... those TE411P cards are in production boxes |
| 15:15.34 | MrChimpy | ns: that's hardly digium's fault |
| 15:15.42 | tzanger | NewSole: well you're in for fun when there is hardware failure.. :-) |
| 15:15.43 | ErMeS|Work | [TK]D-Fender, do you know any third party software to have a past calls list ? |
| 15:15.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:15.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:16.46 | [TK]D-Fender | ErMeS|Work : Past calls to/from where? |
| 15:16.53 | mishehu | I do *not* like waking up in the morning to find that anybody who calls me or anybody I call receives *no* audio at all. |
| 15:16.55 | NewSole | thats why they are not being used..... we are using the vega streams.... comes off the PRI now and encoded g729/g723 |
| 15:17.25 | [TK]D-Fender | NewSole : Happy with the Vegas so far? |
| 15:17.33 | NewSole | very acualy.... |
| 15:18.00 | MrChimpy | tzanger: from looking at that vpmsupport thing I guess the echo cancelling hardware is active by default, yes? |
| 15:18.02 | mishehu | NewSole: a friend of mine is working with the vegastreams, and it sounds like he's not really enjoying working with them. |
| 15:18.05 | *** join/#asterisk kimosabe (n=kimosabe@201.133.216.51) |
| 15:18.25 | kimosabe | i havoe qos isuses for my sip adaptor how can i resolve this eficientlly but economically |
| 15:18.26 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 15:18.29 | NewSole | yes cause they have major issues with H323 Flash |
| 15:18.30 | coppice | maybe he's just a miserable sod :-) |
| 15:18.43 | hackeron | any ideas how do diagnose my problem? - asterisk is just taking 99.9% cpu (0.04% before) with 2 channels all of the sudden |
| 15:18.52 | mishehu | NewSole: I heard they were also having CID problems too. |
| 15:18.58 | NewSole | yup... |
| 15:19.03 | [TK]D-Fender | NewSole : You could always get a TDM or X100P just for timing. |
| 15:19.07 | NewSole | CID un US/Canada |
| 15:19.10 | tzanger | MrChimpy: correct. If a VPM is found it is enabled |
| 15:19.21 | MrChimpy | tzang: cool, thanks |
| 15:19.26 | NewSole | but there is a work around |
| 15:19.37 | donnib | Does anybody why version 1.2.2 critical ? |
| 15:19.38 | NewSole | we are using it... and it works great |
| 15:19.40 | donnib | is critiacl |
| 15:19.42 | *** join/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:19.50 | tzanger | NewSole: can you explain this 20 call limit with ztdummy? |
| 15:19.55 | MrChimpy | i should disable them to see if the $1000 was worth it, but hey, digium deserve money - especially when it isn't mine :) |
| 15:19.57 | LostFrog | I am seriously confused. |
| 15:19.58 | kimosabe | does sipura qos adator qork fine |
| 15:20.21 | LostFrog | An echo test from my phones in India to the PBX in the US didn't have any echo. |
| 15:20.34 | graphyx | If I have previously been running asterisk using the ztdummy and I want to put in a zaptel card, do I need to remove some of the ztdummy stuff before the T1 card will work? |
| 15:20.37 | LostFrog | Just calls terminated to an analog line or another SIP phone. |
| 15:20.56 | LostFrog | I meant didn't have any static. |
| 15:20.57 | graphyx | I got the card in and it shows with the zttool but I can't do a ztcfg without an error in the zaptel.conf file. |
| 15:21.11 | NewSole | after 20 calls where box was converting from IAx to sip... some of the audio would degrade and warble |
| 15:21.21 | donnib | anyone `? |
| 15:21.32 | NewSole | expecialy when transcoding |
| 15:21.43 | tzanger | hmm |
| 15:22.28 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 15:22.47 | NewSole | now we have about 1-200 going on at any given time in day..... and works great.... |
| 15:22.58 | donnib | someone must know why version 1.2.2 is critical and we need a new update !!! |
| 15:23.01 | NewSole | thats per box |
| 15:23.26 | NewSole | donnib> there is an audio timebomb in 1.2.2 |
| 15:23.53 | NewSole | all audio stopped as of last night on sip and iax |
| 15:24.25 | donnib | NetSole: and that means that the sound doesn't work or ??? |
| 15:24.59 | donnib | i have version 1.2.1 right now and i wanted to upgrade to 1.2.2 (asterisk@home made an update for 1.2.2 but not for 1.2.3) |
| 15:25.12 | hackeron | anyone at all? |
| 15:25.16 | hackeron | :( |
| 15:25.20 | Grubs | then wait until A@H comes out with the new 1.2.3 |
| 15:25.36 | Seldon1975 | hello |
| 15:25.45 | Seldon1975 | My SIP users are getting CrossTalk |
| 15:25.57 | Seldon1975 | has anyone else experienced CrossTalk with Asterisk? |
| 15:26.04 | Seldon1975 | it shouldn't be possible |
| 15:26.10 | Seldon1975 | in my understanding |
| 15:26.17 | Seldon1975 | but it's happening |
| 15:26.29 | Peste | hi all... can somebody tell me whats wrong? -- Executing Dial("SIP/100-2d25", "Zap/g1/0011114") in new stack |
| 15:26.29 | Peste | Jan 25 17:25:57 NOTICE[8588]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 15:26.35 | donnib | Grubs: ok. do you know how difficult it is to upgrade from 1.2.1 to 1.2.3 ? i mean i might be able to do it myself without Asterisk@home |
| 15:26.59 | zamsler | hmm |
| 15:26.59 | dudes | Peste - do you have a group assigned in zapata.conf |
| 15:27.07 | Peste | yes |
| 15:27.13 | Peste | group1 for all |
| 15:27.43 | zamsler | I have a dumb question.. Why did the audio issue just pop up? I haven't done any updates in a month??? |
| 15:27.55 | zamsler | and it worked yestrday. |
| 15:28.52 | dudes | Cause sometimes things happen in the future and it doesn't affect you until it gets to that time |
| 15:28.53 | buzzyd | what does it mean when you see Stale nonce received from a client that normally logs in fine? |
| 15:28.59 | zamsler | ahh nvmind |
| 15:29.55 | dudes | Katty - it wants to see how long you'll wait. |
| 15:29.57 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:30.31 | LostFrog | Your phone is physically looking for the person. |
| 15:30.44 | Seldon1975 | is it a Polycom 501? |
| 15:31.10 | hackeron | Seldon1975: are polycom 501 phones bad? -- thats what we're about to order |
| 15:31.21 | Seldon1975 | No, they seem quite good |
| 15:31.33 | Seldon1975 | I just know that Katty is a Polycom user |
| 15:31.57 | Seldon1975 | From what I can tell they're the most popular choise for * users |
| 15:32.01 | Seldon1975 | choice |
| 15:32.11 | hackeron | Katty: you've made a typo, Katy is spelt with 1 't' :) |
| 15:32.22 | Katty | ... |
| 15:32.29 | Katty | stop trying to fix my phone. |
| 15:32.29 | Katty | kthx. |
| 15:32.35 | Katty | shoo, run along and be geeks. |
| 15:32.52 | buzzyd | any ideas what this is about "chan_sip.c:4836 check_auth: Stale nonce received from" |
| 15:33.04 | kimosabe | is any one running sipura 2100 with qos |
| 15:34.19 | kimosabe | does the 2100 sippura suport qos and can i connect it with out having a dsl router menaning i just open the line from my wifi to the router |
| 15:36.40 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 15:38.35 | zoa | yes |
| 15:38.38 | zoa | its keithy |
| 15:38.52 | zoa | install spell checker |
| 15:39.20 | *** join/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 15:39.23 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:39.27 | *** part/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 15:39.29 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 15:39.44 | *** join/#asterisk evilrabbi (i=evilrabb@71.144.78.244) |
| 15:39.50 | evilrabbi | has anyone notice a wierd beeping noise when they place or receive calls? |
| 15:39.54 | fourcheeze | who is Cresl1n and why is it important that we celebrate their birthday? |
| 15:39.57 | evilrabbi | it appears to only be on the voip side |
| 15:40.08 | evilrabbi | the person calling in can't hear it |
| 15:40.27 | zoa | because its all about creslin1 |
| 15:40.42 | zoa | Creslin is the evil matts' twin brother |
| 15:40.57 | evilrabbi | is asterisk 1.2.1 effected by the same bug as 1.2.2? |
| 15:41.04 | zoa | no |
| 15:41.08 | fourcheeze | zoa: I know little of this evil Matt |
| 15:41.08 | |vinsik| | Guys i need some help.. |
| 15:41.15 | *** part/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 15:41.23 | |vinsik| | Im pretty new to this so bear with me. |
| 15:41.26 | zoa | matt is the zaptel developper |
| 15:41.38 | |vinsik| | Ok. |
| 15:41.41 | fourcheeze | ahh ok |
| 15:41.51 | Seldon1975 | arent there several Zaptel devs called Matt? |
| 15:42.09 | fourcheeze | that's how they seem to do so much work |
| 15:42.11 | zoa | nopez |
| 15:42.20 | fourcheeze | so Matt has a brother called Creslin |
| 15:42.21 | zoa | matt fredrickson is the only zaptel developper afaik |
| 15:42.24 | zoa | no no |
| 15:42.27 | zoa | matt is creslin |
| 15:42.31 | fourcheeze | even more ecary |
| 15:42.33 | fourcheeze | scary |
| 15:42.41 | |vinsik| | My wlan phones have probelms connecting to each other. using zyxel phones and linksys wlan stations |
| 15:42.47 | |vinsik| | anybody had problems with those |
| 15:42.49 | fourcheeze | given that we know it's matt's birthday |
| 15:42.49 | |vinsik| | ? |
| 15:42.56 | fourcheeze | and there are 438 peple in the room |
| 15:43.03 | brad_mssw | btw, think there's a typo in the changelog ... should be bug 6349 not 6439 |
| 15:43.09 | fourcheeze | someone can work out the odds of one of them sharing his birthday |
| 15:43.16 | zoa | yes |
| 15:43.23 | zoa | like his evil twin |
| 15:43.37 | fourcheeze | it's quite liely his evil twin would share his birthday |
| 15:43.42 | fourcheeze | but not necessarily so |
| 15:43.45 | zoa | the odds are quite high that they have the same birtdhay |
| 15:43.54 | zoa | unless they are born around midnight |
| 15:44.52 | fourcheeze | zoa: http://www.channel4.com/science/microsites/S/science/body/bodyshock_twin.html |
| 15:45.13 | arkanis | is it possible to have one number (100) which all users can call for their voicebox? |
| 15:45.43 | fourcheeze | arkanis: yes |
| 15:45.59 | fourcheeze | unless by voicebox you mean larynx |
| 15:46.00 | kshumard | arkanis, call VoiceMailMain without a mailbox |
| 15:46.21 | arkanis | can you explain? |
| 15:46.38 | brad_mssw | should probably mention bug #6349 in the topic of this channel as the reason for the 1.2.2->1.2.3 release |
| 15:46.39 | kshumard | exten => 100,1,VoiceMailMain |
| 15:46.58 | TMirage | exten => 1233,2,VoicemailMain() |
| 15:47.06 | TMirage | something like that |
| 15:47.27 | TMirage | then it will ask for mailbox number and pass if given |
| 15:47.29 | nettie | Hi guys.. just a question.. anyone know if there's a "virtual" fax/modem for windows available? Something that will be registered as an extension in asterisk and seen by the client OS like modem which I could use to fax using Outlook fax services ?.. any idea please? |
| 15:47.45 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 15:48.00 | arkanis | ok, cool, but it is possible to have this one number and the user gets redirected directly to his voicebox? |
| 15:48.28 | TMirage | maybe if his voicebox will be the same as the phone number you can make it variable you pass with it |
| 15:49.05 | arkanis | how can I do that (sorry Iam really an newbie) |
| 15:50.31 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:50.32 | TMirage | maybe better read this first http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 15:51.44 | brad_mssw | nettie: seriously doubt you'll find such a thing |
| 15:52.11 | Hmmhesays | propietary shit is so much fun |
| 15:52.34 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 15:54.17 | _4d4m_ | arkanis: call voicemailmain and pass it $CALLERIDNUM |
| 15:54.43 | TMirage | something like this i think |
| 15:54.43 | TMirage | exten => 100,1,VoicemailMain(${calleridnum}) |
| 15:54.49 | *** join/#asterisk peace2u (n=peace2u@219.95.165.118) |
| 15:54.55 | nettie | brad_mssw eheh.. thanx but it would be definitely cool.. you would be able to build and colocate a fax server box whereever you want :) eheh.. |
| 15:55.33 | nettie | brad_mssw: http://board.homeseer.com/printthread.php?t=105382 |
| 15:55.34 | brad_mssw | nettie: probably be possible with a mix of spandsp/txfax and a sip client library |
| 15:55.52 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:56.08 | Blackthorn | Hello All. Does anyone know if nufone is having some troubles at this time? |
| 15:56.12 | *** join/#asterisk Mark33_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 15:57.23 | peace2u | hi guys |
| 15:57.30 | iDunno | good evening. |
| 15:57.37 | peace2u | i seriously in need of experts help |
| 15:57.47 | arkanis | @Tmirage thx |
| 15:58.07 | peace2u | it's regarding heavy load on te411p which causes hang on the whole system |
| 15:58.21 | *** part/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:58.32 | iDunno | stop putting so many bricks on it then. |
| 15:58.57 | zoa | what is causing the heavy load on the te411p ?? |
| 15:59.33 | peace2u | i mean i got heavy IVR traffic coming in |
| 15:59.51 | peace2u | ok...let me explain bout my configuration |
| 16:00.02 | peace2u | i'm actually running autodialer services |
| 16:00.07 | peace2u | i've got PRI's |
| 16:00.15 | peace2u | 2 e1's for incoming |
| 16:00.23 | peace2u | and another 2 e1's for outgoing |
| 16:00.52 | peace2u | when the number of calls coming reach round 20... |
| 16:01.02 | peace2u | it starts to give problem |
| 16:01.20 | peace2u | and it block the whole channels |
| 16:01.30 | peace2u | !! Got reject for frame 107, retransmitting frame 108 now, updating n_r! |
| 16:01.38 | peace2u | once i see this on my screen |
| 16:01.53 | ManxPower | peace2u, what version of Asterisk? |
| 16:01.53 | ManxPower | Are you transcoding? |
| 16:01.56 | peace2u | the processor spikes up to min 30% constantly |
| 16:01.59 | peace2u | 1.2.2 |
| 16:02.08 | peace2u | and now i'm installing 1.2.3 |
| 16:02.12 | NewSole | 20 seems to be magic number |
| 16:02.21 | ManxPower | peace2u, what about 1.2.1? |
| 16:02.31 | peace2u | never tried 1.2.1 |
| 16:03.06 | brad_mssw | ever have success with any version of asterisk ? |
| 16:03.14 | brad_mssw | or have you only tried 1.2.2 ? |
| 16:03.26 | peace2u | yeah i've only tried it with 1.2.2 |
| 16:03.36 | peace2u | and i will constantly get this message too |
| 16:03.37 | peace2u | Jan 25 14:44:00 NOTICE[3696]: chan_zap.c:8180 pri_dchannel: PRI got event: HDLC |
| 16:03.37 | peace2u | Abort (6) on Primary D-channel of span 1 |
| 16:03.38 | ManxPower | This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. |
| 16:03.38 | ManxPower | If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. |
| 16:03.46 | ManxPower | that's a message from the mailing list. |
| 16:03.58 | ManxPower | peace2u, that message is not related to THIS issue. |
| 16:04.05 | brad_mssw | yeah, bug #6349 for anyone who wants to look at it |
| 16:04.45 | peace2u | i think it's more to either zaptel or libpri problem |
| 16:05.29 | *** join/#asterisk ferrency (n=alan@pit27.pair.com) |
| 16:05.39 | NewSole | MARK FOUND THE PROBLEM.... WHAT ABOUT ALL OF US ONLINE AT 3AM REPORTING PROBLEM |
| 16:06.27 | *** join/#asterisk felixD (n=felix@local.epyron.de) |
| 16:06.33 | evilrabbi | NewSole i think he ment mark found the problem in the code and fixed it not found the problem initially |
| 16:06.46 | file | you picky picky people |
| 16:06.51 | tzanger | NewSole: ... what precisely is your complaint? |
| 16:06.55 | tzanger | it wasn't fixed quickly enough? |
| 16:07.11 | peace2u | guys |
| 16:07.17 | peace2u | do u think 1.2.3 will solve my problem |
| 16:07.38 | peace2u | i've spoke to the local distributor for support on my card |
| 16:07.46 | evilrabbi | try it |
| 16:07.50 | peace2u | he recommended me to downgrade it to 1.0 version |
| 16:08.10 | peace2u | i need recommendation here |
| 16:08.12 | tzanger | peace2u: what's the issue? 1.2.3 fixed a very specific problem that manifested itself this morning |
| 16:08.13 | evilrabbi | if it doesnt work then to what the distributor recommended |
| 16:08.37 | asteriskmonkey | what time was the src updated? |
| 16:08.41 | peace2u | my @ss got screwed hard by the customer |
| 16:08.41 | Katty | hmm. |
| 16:08.57 | kimosabe | does any one know whear iu can download the firmwear for wrt54g so that i can do qos with vo/ip |
| 16:10.11 | tzanger | kimosabe: dd-wrt, sveasoft... there are lots. |
| 16:10.29 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 16:10.56 | Peste | hello again! whats wrong with my configs if 'zap show channels' just returns pseudo |
| 16:11.16 | Blackthorn | kimosabe: http://www.sveasoft.com/ |
| 16:12.32 | Peste | no ideas? |
| 16:12.54 | peace2u | have anyone knows wat is the benchmark for zap channels on *-1.2 |
| 16:13.06 | *** join/#asterisk zzxxcc (n=zzxxcc@221.235.45.176) |
| 16:13.12 | Katty | wow, that didn't parse. |
| 16:13.14 | *** join/#asterisk beefsalad (i=crash3m@unaffiliated/crash3m) |
| 16:13.23 | kink0 | well, I check jumpers on my TE405 and are ok (E1) but I had RED alert |
| 16:13.25 | *** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us) |
| 16:14.06 | kink0 | Peste: what reports a zap show status ? |
| 16:14.40 | jarrod | anyway to set the pri in a down state from asterisk cli |
| 16:15.18 | ManxPower | If you get the message "unable to create zap" that means that 1) you installed Asterisk before you installed zaptel, or 2) the kernel modules for your card are not loaded, or 3) your Dial line is wrong. |
| 16:15.26 | ManxPower | jarrod, no |
| 16:15.32 | Peste | Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 |
| 16:15.40 | *** join/#asterisk [|didosch|] (i=_Kardele@213.186.176.80) |
| 16:15.54 | Seldon1975 | Hi, we have a TDM2422E and we're using 1.2 of the software; several users on my SIP network have been experiencing _crosstalk_ with other SIP lines. I didn't think this was possible on SIP lines - anyone know how this can be happening? |
| 16:15.57 | jarrod | how can i set a PRI in a failed state |
| 16:16.02 | ManxPower | Peste, The drivers are loaded then. |
| 16:16.03 | jarrod | without stopping asterisk |
| 16:16.25 | Peste | but i don't get pri workin :( |
| 16:16.26 | ManxPower | jarrod, you mean disable the line? |
| 16:16.48 | ManxPower | Peste, what is the Dial line? |
| 16:17.03 | ManxPower | Peste, you installed Zaptel before you installed Asterisk? |
| 16:17.27 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 16:17.40 | jarrod | manxpower: whatever terminology means to make a PRI plugged into a digium card go down |
| 16:17.46 | jarrod | so calls cant come in |
| 16:17.46 | Peste | i installed zaptel before asterisk |
| 16:17.53 | Peste | my dial line? |
| 16:18.06 | ManxPower | jarrod, see help zap destroy in the CLI. |
| 16:18.16 | ManxPower | Peste, the line in extensions.conf that executes Dial |
| 16:18.54 | ManxPower | Seldon1975, Define "crosstalk". Do you mean "echo" or actual "crosstalk" |
| 16:18.57 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:19.13 | jarrod | will that just drop the current call or destroy the channel until i 'no destroy' it |
| 16:19.17 | Peste | http://pastebin.com/pastebin.php?dl=522465 |
| 16:19.20 | *** join/#asterisk saftsack (n=affe@p54A7F47C.dip.t-dialin.net) |
| 16:19.42 | saftsack | Jan 25 17:15:34 WARNING[793]: chan_misdn.c:3433 cb_events: Got Unknown Event |
| 16:19.49 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 16:19.56 | saftsack | this is what i get, when i send a fax over misdn with iaxmodem |
| 16:19.58 | jarrod | the docs are not clear |
| 16:20.48 | Katty | heh |
| 16:21.32 | kink0 | No D-channels available! what is wrong now ? |
| 16:21.33 | ManxPower | Peste, do you have a group= before the channel= lines in /etc/asterisk/zapata.conf? |
| 16:21.42 | Mark33_Halverson | good morning all - does this patch / problem have anything to do with NAT? SIP <-> SIP bridging stopped for me through NAT, but work on public IP |
| 16:21.57 | peace2u | can i assume that the te410p is faulty when i load the wct4xxp the processor usage hike up to constant 30-40% |
| 16:22.16 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:22.32 | Peste | yes i have http://pastebin.com/522473 |
| 16:22.32 | tzanger | "It is a fascinating material, created by a sperm whale's intestine in response to irritation, probably caused by the undigested beaks of squid. The waxy mass is coughed out by the whale during a belch which is reported to be audible kilometers away, and is a putrid stinking mass." |
| 16:22.37 | tzanger | *giggle* |
| 16:22.54 | brad_mssw | Mark33_Halverson: possibly, look at bug 6349 |
| 16:22.54 | rob0 | tzanger: save some for me! |
| 16:22.56 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:22.57 | a1fa | tada |
| 16:22.58 | a1fa | :P |
| 16:22.59 | gnosys | can someone point me to some details regarding the critical update from 1.2.2 to 1.2.3? I have a bizarre bug in 1.2.2 that I can't track down and I wonder if this patch is the cause. I'm upgrading to 1.2.3 now, but am interested in the details anyway... |
| 16:23.00 | a1fa | sup dawg |
| 16:23.13 | Mark33_Halverson | yeah i did...mostly says zap <-> sip bridge...so wasn't sure |
| 16:23.15 | a1fa | lol |
| 16:23.17 | a1fa | suckah.. |
| 16:23.20 | brad_mssw | ok, can someone freaking put a reference to bug 6349 in the topic |
| 16:23.41 | Mark33_Halverson | callsign? |
| 16:23.49 | *** join/#asterisk SDGL (n=sdgl@64.5.206.131) |
| 16:23.57 | warthawg | K1GPL |
| 16:24.13 | brad_mssw | Mark33_Halverson: try it, i don't know if anyone fully knows the consequences besides markster |
| 16:24.20 | Mark33_Halverson | lol |
| 16:24.22 | Mark33_Halverson | ok |
| 16:24.33 | sivana | ~seen JerJer |
| 16:24.38 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 6d 18h 53m 44s ago, saying: 'thanks again'. |
| 16:24.58 | Mark33_Halverson | kc6rlb |
| 16:25.12 | _Sam-- | darwin_35 you there? |
| 16:25.20 | arkanis | according to voip-info.org I can use "t" when I call "dial", so the called user can transfer the call |
| 16:25.20 | Peste | ManxPower, kink0: any ideas? |
| 16:25.22 | *** join/#asterisk muzzz_ (n=chatzill@218.111.65.17) |
| 16:25.46 | arkanis | so does that mean I can hit # in my softphone und transfer the call to another user? |
| 16:26.28 | saftsack | Jan 25 17:24:43 WARNING[796]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host 127.0.0.1 on IAX2/iaxmodem-4 (type = 6, subclass = 3, ts=22008, seqno=11) |
| 16:26.30 | warthawg | i am off the air at the moment, waiting for the cement to be poured for a small tower. just do cw anyway. |
| 16:26.48 | [TK]D-Fender | arkanis : Yes. |
| 16:26.57 | ManxPower | jarrod, you can simply disconnect a call see softhangup, or you can destroy the channel so that you have to unload and reload asterisk. |
| 16:27.23 | ManxPower | Peste, if you do a "pri debug" and wait 2 mins, do you see debug info? |
| 16:27.43 | ManxPower | If you do not see the debug info that call up your telco and say "we have no traffic on the D-Channel" |
| 16:28.03 | *** join/#asterisk svenna_ (n=svenna@p548D278E.dip0.t-ipconnect.de) |
| 16:28.30 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 16:28.47 | ManxPower | saftsack, sounds like you are trying to dial localhost |
| 16:29.08 | saftsack | ManxPower, oh that was a silly errormessage because i killed iaxmodem |
| 16:29.13 | zoa | sam, i am here :) |
| 16:29.26 | Peste | no such command.. oO |
| 16:29.39 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:29.42 | ManxPower | gnosys, if you were on the mailing lists you would know. 8-) |
| 16:29.59 | ManxPower | Peste, "pro debug span X" where X is the span of your PRI |
| 16:30.03 | ManxPower | ..er. |
| 16:30.04 | _Sam-- | zoa: if you could fix darwin_35's router at rockynet it would be most appreciated :) |
| 16:30.06 | ManxPower | Peste, "pri debug span X" where X is the span of your PRI |
| 16:30.11 | gnosys | woops... I'll go read. sorry to bother folks... |
| 16:30.23 | saftsack | ManxPower, oder warte ma, kann gut sein |
| 16:30.25 | a1fa | it took me 1 min to update to 1.2.3 |
| 16:30.28 | Peste | no pri running on span 1 :( |
| 16:30.29 | a1fa | :P |
| 16:30.32 | zoa | haha |
| 16:30.36 | ManxPower | if you still don't have the command then stop, rebuild and reinstall in this order: zaptel, libpri, asterisk |
| 16:30.37 | zoa | i can do everything ! |
| 16:30.43 | _Sam-- | i bet you could too |
| 16:30.46 | zoa | all expenses paid ? :) |
| 16:30.55 | zoa | how broken is it |
| 16:31.00 | zoa | can we remotely login to it ? |
| 16:31.09 | _Sam-- | same as usual, just really small, annoying packet loss (1-3%) |
| 16:31.20 | zoa | cisco ? |
| 16:31.36 | _Sam-- | i hope so....darwin_35 = tech guy at teliax |
| 16:31.38 | Seldon1975 | ManxPower: Actual cross-talk - user A can hear user B's conversation, somehwat ditorted |
| 16:31.40 | zoa | aaah |
| 16:31.40 | zoa | k |
| 16:31.50 | Peste | i have the command, but it tells me that i have no spans running or no pri on span |
| 16:31.51 | zoa | so they probably wont let me on it then :) |
| 16:32.00 | ManxPower | Seldon1975, I think your users are on drugs |
| 16:32.01 | _Sam-- | at what point does packet loss become noticeable? |
| 16:32.11 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 16:32.11 | zoa | immediately |
| 16:32.11 | _Sam-- | it seems like 1% my clients dont notice, but 3% they do |
| 16:32.19 | arkanis | @d-fender: so, when I recieve a call I can dial # and the usernumber and the call gets redirected? |
| 16:32.24 | Peste | do u speak german? |
| 16:32.24 | zoa | it could come in series |
| 16:32.29 | Seldon1975 | ManxPower: I know it sounds weird - BUT I HAVE ACTUALLY HEARD IT |
| 16:32.33 | zoa | i spreche ein bitchen deutsh |
| 16:32.35 | zoa | or something |
| 16:32.39 | Peste | nic |
| 16:32.40 | Seldon1975 | ich auch |
| 16:32.45 | zoa | ich bin ein berliner |
| 16:32.49 | Peste | :D |
| 16:32.56 | zoa | das ist toll! |
| 16:32.56 | Peste | ich bin österreicher |
| 16:33.01 | zoa | woher geht der bus ? |
| 16:33.40 | Seldon1975 | THis is really frustrating; all you Asterisk people wont believe that I'm hearing crosstalk on my SIP Phones; I wouldn't believe it either until I heard it |
| 16:33.42 | _Sam-- | zoa: how hard would it be to make a hard phone, now that you have the idefisk? could you apply any of that towards a hard phone? |
| 16:33.43 | [TK]D-Fender | Seldon1975 : I have had cross-talk a few times (extremely rare) in my customer service dept.... |
| 16:33.50 | Seldon1975 | ok |
| 16:33.53 | Seldon1975 | thank god |
| 16:34.06 | Seldon1975 | my users get it more than rarely |
| 16:34.07 | [TK]D-Fender | Seldon1975 : Not crazy... its happened. I'm running Polycom IP 600's there |
| 16:34.26 | Seldon1975 | ok we have PCom 601s, 501s and 301s |
| 16:34.30 | [TK]D-Fender | Seldon1975 : do you have a lot of them daisy-chained together? |
| 16:35.01 | Seldon1975 | a lot of them are connected to simple hubs |
| 16:35.09 | [TK]D-Fender | Seldon1975 : The only people here that got that were all tied in a chain from LAN/2nd eth port. |
| 16:35.11 | Mark33_Halverson | get a good commerical switch...running 300s on 3com superstack switchs no problems |
| 16:35.12 | Seldon1975 | which branch off the main network Switch |
| 16:35.19 | [TK]D-Fender | Seldon1975 : I might suspect the hubs... |
| 16:35.33 | Seldon1975 | ok we have a 3 Com superswitch |
| 16:35.43 | Mark33_Halverson | hubs would be a prob |
| 16:35.45 | Seldon1975 | but cheaper hubs at each group of phones |
| 16:35.47 | hans | crosstalk? as in hearing part of another conversation? |
| 16:35.51 | Seldon1975 | yes |
| 16:35.53 | Seldon1975 | exactly |
| 16:35.55 | zoa | sam, no it would be starting over |
| 16:36.50 | hans | in sip that would only be possible if a phones picks up an sdp packet that wasn't intended for it, right? |
| 16:36.51 | Seldon1975 | ok so we suspect the small 8-port netgear hubs might be the issue? |
| 16:36.54 | hans | is there a NAT involved? |
| 16:37.17 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfim6.dialup.mindspring.com) |
| 16:37.17 | Peste | do i need a restart after recompiling? |
| 16:37.37 | Seldon1975 | no |
| 16:37.45 | hans | s/sdp/rtp/ |
| 16:38.05 | _Sam-- | i have an idea for asterisk...instead of sip and iax peers being seperate...why couldnt you just have one peers file, and in that config you could specify which protocol(s) that peer can use? |
| 16:38.10 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfim6.dialup.mindspring.com) |
| 16:38.16 | _Sam-- | or friends |
| 16:38.21 | zoa | _Sam--, you can do that with realtime |
| 16:38.31 | Seldon1975 | can any of you propose a solution; other than removing the Hubs that is :} |
| 16:38.33 | zoa | the configs are not completelyt he same |
| 16:38.45 | _Sam-- | i have realtime but still have two tables |
| 16:38.47 | _Sam-- | one for sip peers |
| 16:38.49 | _Sam-- | and one for iax peers |
| 16:38.56 | _Sam-- | or buddies , whatever you want to call it |
| 16:39.06 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 16:39.41 | hans | Seldon1975: so you suspect the hub is transmogrifying the rtp packets? |
| 16:39.55 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 16:40.01 | Seldon1975 | well thats what the guys here are suggesting |
| 16:40.05 | frenzy | hey all |
| 16:40.06 | Seldon1975 | I have no ideas! |
| 16:40.09 | Peste | <PROTECTED> |
| 16:40.13 | frenzy | what does 603 mean ? |
| 16:40.26 | *** join/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 16:40.38 | hans | Seldon1975: is there a NAT device in the path? |
| 16:40.48 | Seldon1975 | hans: no |
| 16:41.02 | Seyr | Anyone know why I keep getting the message "Attempting native bridge" every time I press * when I am on a call? |
| 16:41.06 | hans | is the crosstalk constant through a call, or sporadic? |
| 16:41.17 | *** join/#asterisk chrismm (n=c@dslb-084-057-240-043.pools.arcor-ip.net) |
| 16:41.21 | Seyr | trying to do _*1xx in a diaplan so I can transfer a call to voicemail |
| 16:41.30 | Seldon1975 | hans: it seems constant |
| 16:41.32 | frenzy | Seyr: I think asterisk is transcoding |
| 16:42.20 | hans | Seldon1975: so from the start of the call to the end of the call? |
| 16:42.24 | Peste | still now zap channels or pri spans |
| 16:42.44 | *** part/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 16:42.58 | *** join/#asterisk fugitivo (n=ajf@201.255.176.51) |
| 16:43.05 | Seldon1975 | hans: yes; the whole call |
| 16:43.20 | hans | Seldon1975: every call, or only occasional calls? |
| 16:43.34 | Seyr | So anyone know what I can do to make pressing * during a call work? |
| 16:43.44 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 16:43.50 | hans | does the other call (the one that's being overheard) also have the crosstalk? |
| 16:43.57 | *** join/#asterisk TrickyR (n=MT@157.246.8.43) |
| 16:44.00 | chiardon | hello |
| 16:44.13 | arkanis | d-fence: So can I just press # and the sip-account to which I want to transfer (redirect) the call? |
| 16:44.21 | fugitivo | Seyr: are you using atas or ip phones? |
| 16:44.38 | Seyr | 7960 |
| 16:44.55 | Blackthorn | Hello All. Does anyone know if nufone is having some troubles at this time? |
| 16:44.59 | a1fa | anybody have sploit for 1.2.2 :P |
| 16:45.12 | fugitivo | 0 dayz |
| 16:45.28 | Seldon1975 | hans: I'll just check witht he users on that |
| 16:45.52 | a1fa | fugitivo : now it is the time pwn somebody |
| 16:45.57 | chiardon | someone could help me to know where i configure addons |
| 16:46.05 | Damin | Blackthorn: I noticed that my home 800 number doesn't work through them.. |
| 16:46.07 | a1fa | 31337! |
| 16:46.30 | fugitivo | 1337 r0x0r |
| 16:46.44 | a1fa | r0x0r my s0x0rs! |
| 16:46.45 | a1fa | :p |
| 16:46.49 | fugitivo | lol |
| 16:46.57 | a1fa | yeah dude :P |
| 16:46.59 | a1fa | i just patched my ass ;P |
| 16:47.06 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:47.06 | fugitivo | hehe |
| 16:47.12 | Blackthorn | ij yes guesst they are down. I just called there telephne number and it goes to a "your party does not answer" message. |
| 16:47.14 | a1fa | i am good to go :p |
| 16:47.29 | hans | Seldon1975: since we're talking packets here, I seriously doubt it's a HUB problem. A switch _might_ be feasibly a problem, but even that is unlikely. something that rewrites packets, like a NAT device or some fancy router would be a primary suspect. Or bad devices that are picking up packets that don't have their IP addresses (does this only happen on one kind of phone? what about a different phone or softphone) |
| 16:47.36 | *** join/#asterisk roulduke (i=k682lkkv@p508D0C99.dip0.t-ipconnect.de) |
| 16:48.01 | a1fa | i need a .g wifi phone that supports WPA |
| 16:48.01 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 16:48.02 | hans | or, maybe asterisk is sending out the packets to both sip devices, but that's unlikely too. Ethereal is your friend |
| 16:48.14 | a1fa | hans : you tripping |
| 16:48.19 | a1fa | hans : tcpdump is your friend |
| 16:48.31 | hans | :) |
| 16:48.38 | hans | some packet sniffer, anyhow |
| 16:48.40 | asteriskmonkey | hey what revisions is considers 1.23? |
| 16:48.52 | *** join/#asterisk gongoputch (n=gongoput@pcp01486721pcs.limstn01.de.comcast.net) |
| 16:49.00 | a1fa | hans : what.. you cant be elite haxor, and not use tcpdump, |
| 16:49.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:49.23 | FuriousGeorg1 | people are complaining that inbund calls from cellphones tend to get cut off (this is not a joke) same with outgoing to toll free via iax2 provider |
| 16:49.24 | hans | a1fa: why G? |
| 16:49.26 | pifiu | hey everyone wasup |
| 16:49.36 | a1fa | hans : ok .b that supports WPA |
| 16:49.47 | Seldon1975 | hans: ok thanks - the users say its happening every time they are on the phone at the same time as the person they get crosstalk with |
| 16:49.48 | hans | ah, WPA :) |
| 16:49.53 | asteriskmonkey | ive notices huge issues with link2voip in terms of quailty and drops |
| 16:49.56 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:50.10 | pifiu | hey everyone wasup |
| 16:50.11 | hans | Seldon1975: that's good news - it means you can probably solve it by tweaking the phone configs |
| 16:50.23 | asteriskmonkey | wheat VERSION is considered 1.23? |
| 16:50.23 | hans | maybe change the rtp port ranges, or IP addresses |
| 16:50.37 | Seldon1975 | hans: ok thanks |
| 16:50.42 | hans | in fact, you ought to check to see that they have distinct IPs and/or mac addresses |
| 16:50.46 | a1fa | http://www.voipsupply.com/product_info.php?products_id=1067 |
| 16:50.47 | brad_mssw | asteriskmonkey: what? |
| 16:50.54 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:50.56 | a1fa | i dont know if it supports WPA :( |
| 16:50.57 | Seyr | ok, I can press # during a call (to transfer) but I cant press * |
| 16:50.59 | [TK]D-Fender | a1fa : lONG TIME NO SEE... HOW'S YOUR SETUP WORKING? |
| 16:51.00 | brad_mssw | asteriskmonkey: are you asking what changed in asterisk 1.2.3 ? |
| 16:51.10 | trixter | etel is interesting... at least food is included |
| 16:51.13 | trixter | but no open bar :( |
| 16:51.17 | [TK]D-Fender | a1fa : And whay are you looking for a WIFI phone? |
| 16:51.17 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 16:51.17 | asteriskmonkey | brad_mssw: no i know what changed i want to make sure i put the latest one in |
| 16:51.20 | Seldon1975 | hans: they are useful suggestions; Ill take a look |
| 16:51.34 | Peste | anyone have a clue how to create a span and with pri running on it? |
| 16:51.35 | [TK]D-Fender | pifiu : So, how's that IAX2 link working? |
| 16:51.37 | Seldon1975 | hans: thanks |
| 16:51.43 | a1fa | [TK]D-Fender : excelent.. i just upgraded to 1.2.3 |
| 16:51.47 | asteriskmonkey | brad_mssw: i upgraded early today and wanted to know version number just incase i screwd it by putting and old on in |
| 16:51.47 | supjigatr | Anyone with a MAXTNT and * seen DTMF work with G711 and not G729a? |
| 16:51.53 | [TK]D-Fender | a1fa : congrats |
| 16:51.58 | a1fa | [TK]D-Fender : i also added voicechanger app :P |
| 16:51.59 | tzanger | supjigatr: don't use inband DTMF |
| 16:52.02 | gnosys | I wonder if I could get a show of hands on how many people use the 1.2 "stable" branch in production? Did the timebomb bug affect only 1.2.2 or earlier releases too? |
| 16:52.05 | brad_mssw | asteriskmonkey: my asterisk console shows : Asterisk 1.2.3, Copyright (C) 1999 - 2006 Digium, Inc. and others. |
| 16:52.09 | a1fa | [TK]D-Fender : had a blast with my friends the other night :P |
| 16:52.10 | hans | Seldon1975: sure, good luck |
| 16:52.18 | brad_mssw | gnosys: just 1.2.2 supposedly |
| 16:52.22 | [TK]D-Fender | a1fa : sure thing... Darth :) |
| 16:52.23 | supjigatr | tzanger: I have the maxtnt set to RTP for both. |
| 16:52.23 | a1fa | gnosys : lol |
| 16:52.34 | asteriskmonkey | mine show Asterisk SVN-branch-1.2-r8632M built by root @ asterisk1.local on a i686 running Linux on 2006-01-25 14:12:27 UTC |
| 16:52.36 | a1fa | [TK]D-Fender : Luke, I am your father! |
| 16:52.45 | tzanger | supjigatr: that's inband. use RTC2833. |
| 16:52.46 | a1fa | oMFG |
| 16:52.48 | tzanger | er RFC2833. |
| 16:52.49 | a1fa | built by root |
| 16:52.49 | a1fa | lolk |
| 16:52.50 | supjigatr | tzanger: So G711 is working but what should both ends be set with G729a? |
| 16:52.52 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 16:52.58 | Seldon1975 | Last time we met I was but the apprentice, now I AM THE MASTER |
| 16:53.06 | supjigatr | tzanger: Its that called something else in the tnt? |
| 16:53.09 | tzanger | supjigatr: DTMF (any continuous tone) is haphazard at best with any compressed voice codec. |
| 16:53.30 | *** join/#asterisk lahaine (n=lahaine@m219.net81-66-61.noos.fr) |
| 16:53.31 | [TK]D-Fender | INBAND = EVIL. |
| 16:53.44 | brad_mssw | asteriskmonkey: tag I see is r8647 for 1.2.3 |
| 16:53.46 | supjigatr | g711 is ok with RTP right? |
| 16:53.56 | tzanger | supjigatr: correct |
| 16:53.59 | kink0 | an stupid confirmation, extreme from the carrier provide is NT/M while my asterisk must be TE/S, right ? |
| 16:54.14 | asteriskmonkey | ah crap, i must have updated before the patch was realease |
| 16:54.24 | supjigatr | tzanger: Any idea what the option in the TNT is for rfc2833? |
| 16:54.38 | tzanger | I think it's actually called "rfc2833" |
| 16:54.44 | asteriskmonkey | quick question also do i have to delete my old app_rxfax.so beofre i recompile another one or will it overright by nature? |
| 16:54.49 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 16:55.06 | supjigatr | I have TSAO 11.0 |
| 16:55.14 | supjigatr | I'll have to check. |
| 16:55.16 | chiardon | my * Box don't go up and the message is related with libmysql client . . .What must i do to resole this?? |
| 16:55.22 | *** join/#asterisk warmcat (n=chatzill@cpc1-nthc5-0-0-cust289.nrth.cable.ntl.com) |
| 16:56.08 | kink0 | people who has connected ther Asterisk T1/E1 cards to the carrier, has her T1/E1 as TE/S , is right ? |
| 16:56.21 | brad_mssw | asteriskmonkey: the fix was actually in r8588 it appears |
| 16:56.35 | brad_mssw | asteriskmonkey: but when tagging 1.2.3, it's official release tag is r8647 |
| 16:56.52 | FuriousGeorge | im having people tell me that inbound calls from cellphones are getting disconnected, and thye say it happens so often it cant be a just bad connection. the only time i experienced it was when driving. |
| 16:57.04 | FuriousGeorge | they say certain calls the person will call back 5 times and get cut off every time |
| 16:57.10 | warmcat | Hi folks, trying to use a SIP phone into an Asterisk running on ARM. It has OSS drivers for /dev/dsp and last night was working as expected :-/ Thismorning, asterisk is not opening /dev/dsp when the call comes in, despite it recognizes the SIP call and says it is handled by OSS/dsp |
| 16:57.26 | *** part/#asterisk beefsalad (i=crash3m@unaffiliated/crash3m) |
| 16:57.28 | brad_mssw | asteriskmonkey: the rxfax shouldn't need to be recompiled, the module API didn't change afaik |
| 16:57.32 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 16:57.44 | warmcat | The SIP phone is sending data back on UDP but Asterisk is not sending anything out, doesn't even have /dev/dsp open |
| 16:58.00 | brad_mssw | warmcat: version of asterisk ? |
| 16:58.13 | asteriskmonkey | brad_mssw: ive never been able to get it working after 1.0.9, i recomiled this morning with no erroes but still get module wont load |
| 16:58.22 | asteriskmonkey | any clue how to debug further? |
| 16:58.30 | warmcat | brad_mssw: 1.2.2 |
| 16:59.13 | brad_mssw | warmcat: look at http://bugs.digium.com/view.php?id=6349 which is fixed in 1.2.3 |
| 16:59.26 | warmcat | ohoho sounds good! |
| 16:59.43 | FuriousGeorge | am i invisible or just being ignored, anyone see this? |
| 16:59.50 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 16:59.51 | brad_mssw | asteriskmonkey: using 0.0.2pre23 of spandsp and rx/txfax ... works fine |
| 16:59.55 | jpk | Probably only ignored. |
| 16:59.57 | warmcat | lol it broke according to the date! |
| 16:59.59 | FuriousGeorge | k, good |
| 17:00.04 | brad_mssw | asteriskmonkey: actually, it's the only version that has worked properly for me, actually |
| 17:00.05 | Peste | anyone have aan idea how to create a span with pri running on it? i use a TE110P card connected with E1 to a AG2000 card. libpri and zaptel are up-to-date and i have green lights, but always get a "Unable to create channel of type 'Zap' (cause 0 - Unknown)" |
| 17:00.11 | ManxPower | FuriousGeorge, your problem is so bizzare I have no comment |
| 17:00.37 | ManxPower | Peste, what happened when you did a pri debug command? |
| 17:00.46 | FuriousGeorge | ManxPower: thats better than being ignored i guess :) |
| 17:00.51 | warmcat | thanks a lot Brad I will update myself |
| 17:01.02 | *** join/#asterisk CosmicRay (n=jgoerzen@2002:452c:8843:1:20e:a6ff:fe66:c5a3) |
| 17:01.07 | Peste | it says no pri running on the span |
| 17:01.18 | ManxPower | Peste, PASTE the actual command you are running? |
| 17:01.46 | CosmicRay | quick dialplan question: how can I "escape" paramenters? I've written a macro that takes an ${ARG2} that should be passed to Goto. But if I pass, say, foo,1,2 , then ${ARG2} is foo, ${ARG3} is 1, etc. |
| 17:01.48 | Peste | pri debug span 1 |
| 17:01.54 | CosmicRay | I tried putting it in quotes but Goto doesn't like that |
| 17:01.55 | knight_ | Can anyone suggestion who I should talk to regarding an illegit company who is using stolen Digium enterprise codecs? |
| 17:02.05 | tzanger | knight_: I'd probably talk to digium |
| 17:02.11 | supjigatr | tzanger: CLI says rtp, inbound and outofband are only options. |
| 17:02.13 | knight_ | teezee, yeah I'm planning on it |
| 17:02.21 | tzanger | supjigatr: well use your head. :-) |
| 17:02.24 | knight_ | I mean specific extensions or people |
| 17:02.31 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 17:02.35 | ManxPower | CosmicRay, try \ before the , I don't think that will work. If it doesn't you'll need to use something other than , |
| 17:02.58 | CosmicRay | ManxPower: I tried using | but had the same problem. |
| 17:03.01 | supjigatr | tzanger: I've tried all the settings. |
| 17:03.03 | CosmicRay | I also tried \ but it didn't help either. |
| 17:03.08 | supjigatr | RTP should be rfc2833 right? |
| 17:03.20 | ManxPower | CosmicRay, , and | are special characters to asterisk |
| 17:03.20 | brad_mssw | CosmicRay: , and | are reserved ... use something like : |
| 17:03.25 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 17:03.32 | CosmicRay | brad_mssw: but Goto will not understand :, will it? |
| 17:03.38 | ManxPower | supjigatr, no rfc2833 is RTP silence supression |
| 17:03.39 | FuriousGeorge | ManxPower: i tried commenting callprogress=yes in zapata.conf, I should be ok with that right? |
| 17:03.58 | ManxPower | FuriousGeorge, callprogress and busydetect will randomly disconnect calls. |
| 17:03.59 | knight_ | I can't seem to find a number on Digium's site |
| 17:04.02 | brad_mssw | CosmicRay: no, thought that was the point ... so you could get a list without it being passed to multiple arguments |
| 17:04.14 | brad_mssw | CosmicRay: maybe I'm not clear on what you're trying to do |
| 17:04.14 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 17:04.15 | CosmicRay | brad_mssw: then how would I convert it to something that Goto could grok? |
| 17:04.21 | ManxPower | CosmicRay, don't fight it, pass Goto multiple args. |
| 17:04.21 | FuriousGeorge | ManxPower: maybe thats what it was then, i guess ill find out |
| 17:04.33 | mogorman | digium's number is 1877linuxme |
| 17:04.36 | stack_ | I'm looking at setting up an Asterisk PBX. We need to get a bunch of VOIP phones, cheap. I'm looking at voipsupply.com right now. Are there any recommendations for the lower end IP phones? |
| 17:04.47 | knight_ | thanks |
| 17:04.48 | *** part/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 17:04.52 | CosmicRay | ManxPower: well, it is convenient to not pass Goto all the args all the time. |
| 17:04.53 | *** join/#asterisk johnjw (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 17:04.55 | CosmicRay | hmm. |
| 17:04.56 | ManxPower | Macro(bob,outgoing,${EXTEN},1) then in the macro Goro(${ARG1},${ARG2},${ARG3}) |
| 17:05.09 | brad_mssw | stack_: the linksys 941 has been great ... the polycom 501 for a little more is probably better |
| 17:05.18 | SPoon_TSX | Good morning everyone. I got some question regarding implement the T1 PRI to my office. |
| 17:05.24 | brad_mssw | stack_: don't go with grandstream gxp2000 unless you want echo issues |
| 17:05.27 | ManxPower | CosmicRay, if you don't do it that way you'll have to do all sorts of work in your macro. |
| 17:05.40 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 17:05.54 | SPoon_TSX | May I know is it TRUE that if I go for T1 PRI route, I should not have any echo issue? (If I use up all 24 channels for voice?) |
| 17:05.57 | stack_ | brad_mssw, there is a Linksys 841 for cheaper... what do you think? |
| 17:06.17 | stack_ | brad_mssw, my boss wants to stay under $100 a phone |
| 17:06.41 | ManxPower | i.e. you'll have to do something like Marco(bob,context^extension^priority) then in the macro break each item out, check to see if it exists, then rebuild the information with commas instead of carrets, then call goto |
| 17:06.47 | brad_mssw | stack_: the 841 sucks ... got one, hate it |
| 17:06.54 | ManxPower | SPoon_TSX, WRONG! |
| 17:06.56 | brad_mssw | stack_: the 941 in no way is similar to the 841 |
| 17:07.14 | ManxPower | If you are on a PRI you can still have echo generated by the FAR END analog lines |
| 17:07.25 | stack_ | brad_mssw, is there anything sub $100 that is worth getting? |
| 17:07.26 | brad_mssw | stack_: well, you get what you pay for ... < $100 is very difficult unless you're buying in volume |
| 17:07.40 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 17:07.41 | ManxPower | no, there is no IP phone for under $100 that is any good. |
| 17:07.42 | brad_mssw | stack_: stick with ATA's and regular phones if you need to be < $100 |
| 17:07.44 | SPoon_TSX | ManxPower: Oh? So i still get the cho issue as Analog line. then? |
| 17:07.52 | ManxPower | that's like asking if there's a computer under $300 that is any good. |
| 17:08.01 | ManxPower | SPoon_TSX, Correct. |
| 17:08.02 | stack_ | gotcha, thanks |
| 17:08.23 | ManxPower | A PRI will eliminate echo caused by your LOCAL line, but the far end is often analog and causes echo. |
| 17:08.54 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 17:09.16 | SPoon_TSX | ManxPower: So what will you suggest to have a carrier grade voice quality including eliminating the echo issue? PRI or just stick with Analog? |
| 17:09.20 | asteriskmonkey | brad_mssw: spandsp-0.0.2pre23.tar.gz |
| 17:09.27 | warthawg | my voip provider is having a lot of problems this morning. anyone have a recommendation for a provider that works well with Asterisk? |
| 17:09.42 | brad_mssw | asteriskmonkey: yeah, that's what I've currently got up and running |
| 17:09.49 | SPoon_TSX | ManxPower: My office going to need 8 lines for the PSTN connection. What will you suggest? |
| 17:09.54 | ManxPower | SPoon_TSX, PRI and hardware echocan like Tellabs or other hardware echocan. You can also try the Digium card with the echocan module. |
| 17:09.59 | SplasPood | Does anyone know if it's possible to disable (ie, prevent the user from setting) the Do Not Disturb function on the Polycom Soundpoint phones? |
| 17:10.11 | ManxPower | SPoon_TSX, I suggest you don't use asterisk for that. |
| 17:10.23 | kink0 | ManxPower, if you connect your asterisk PRI card to the carrier , are you TE/S , right ? |
| 17:10.25 | _Sam-- | warthawg who is your provider? |
| 17:10.32 | warthawg | _Sam--, nufone |
| 17:10.32 | ManxPower | SPoon_TSX, remember you can only year echo if your local phones are IP |
| 17:10.53 | ManxPower | kink0, PRI does not have the concept of S/U |
| 17:11.00 | saftsack | are there any a/d changer for using a normal fax directly with isdn? |
| 17:11.06 | ManxPower | that's a BRI only thing |
| 17:11.22 | kink0 | ManxPower, but you set your PRI card as TE or NT ? |
| 17:11.27 | SPoon_TSX | ManxPower: Oh, you mean I better off from Asterisk if I need 8 PSTN line? What other software you will suggest? |
| 17:11.39 | ManxPower | kink0, PRI DOES NOT HAVE TE or NT |
| 17:11.52 | ManxPower | SPoon_TSX, I suggest a cheap hardware key system. |
| 17:12.17 | asteriskmonkey | brad_mssw: im using spandsp-0.0.2pre23.tar.gz and i cant it loaded :( |
| 17:12.21 | ManxPower | If you really want to use Asterisk, stay away from VoIP and use analog phones and lines with a channelbank. |
| 17:12.32 | kink0 | ManxPower, ahhh ok, I means E1 |
| 17:12.35 | SPoon_TSX | ManxPower: Can you give me some example? |
| 17:12.36 | ManxPower | you won't have to worry about echo then. |
| 17:12.39 | kink0 | sorry for mistake |
| 17:12.39 | brad_mssw | asteriskmonkey: did you compile the latest rxfax and txfax ? |
| 17:12.44 | brad_mssw | asteriskmonkey: loads fine here |
| 17:13.02 | asteriskmonkey | brad_mssw: yes i did , and i got no errors |
| 17:13.05 | warthawg | ManxPower, is that true of all VOIP, like you buy from the cable company? |
| 17:13.07 | stack_ | with regards to fax detection, how well does Asterisk do |
| 17:13.07 | ManxPower | kink0, The only lines that have the idea of NT/TE is BRI. |
| 17:13.12 | *** join/#asterisk elephantMan (n=elephant@252.205.103-84.rev.gaoland.net) |
| 17:13.20 | brad_mssw | asteriskmonkey: what errors are you getting upon load ? |
| 17:13.33 | SPoon_TSX | ManxPower: okay.... too bad, which means all of our existing IP Phone will become a junk?? |
| 17:13.48 | asteriskmonkey | brad_mssw: just says wont load |
| 17:14.00 | *** join/#asterisk enigma___ (n=toxic@shell.franken.de) |
| 17:14.17 | brad_mssw | asteriskmonkey: no idea ... using gentoo, so their e-build built it from source for me ... they may have some patches I'm not aware of |
| 17:14.20 | kink0 | ManxPower, ok, that is what I am asked at my 2N Stargate for the PRI configuration, mode NT/s NT/M ... TE/M or TE/S |
| 17:14.21 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 17:14.26 | gnosys | After the timebomb bug and several hours of lost phone connectivity in my office this morning, I'm considering downgrading from 1.2.x to 1.0.x out of concern that 1.2 is not really stable and that there are other bugs in 1.2.x that are likely to make many headaches for me in the future, but as of 11/21/05, the asterisk.org page says: "It is very likely that this will be the final release of the 1.0 branch of Asterisk. Users are strongly encouraged to |
| 17:14.36 | enigma___ | bristuff-0.3.0-PRE-1i seems to have borked timestamps :-/ |
| 17:14.45 | enigma___ | ( or better its tarball ) |
| 17:15.18 | Peste | after help from kink0: ok... some values where at no in my conf, BUT the real prob was: [channels] showed up in the beginning... the 1000 lines of comments and [channels] showed up again at the end... thank you very much... now pri is working well :) |
| 17:15.42 | ManxPower | kink0, None of those. You have PRI CPI or PRI NETWORK |
| 17:15.51 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 17:15.55 | brad_mssw | gnosys: it's your decision, obviously, but any bugs in 1.0.x, don't expect them to be fixed .... typically it just means don't jump on board to each release immediately, as there may be bugs |
| 17:16.16 | brad_mssw | gnosys: 1.2.2 was only released a few days ago, and it was a bug specific to 1.2.2 |
| 17:16.29 | warmcat | brad_mssw: owe you a beer, it's back working properly :-) Thanks again |
| 17:16.31 | ManxPower | kink0, you are asking what type of petrol an cat requires. Cat's don't require petrol. PRIs don't use NT or TE modes. |
| 17:16.34 | asteriskmonkey | brad_mssw: anyway i can look for more debug info anywhere |
| 17:16.59 | Hmmhesays | i could use a beer right now |
| 17:17.01 | Hmmhesays | and some pizza |
| 17:17.01 | *** join/#asterisk elvisthedj (n=Johnny@216.220.30.20) |
| 17:17.02 | kink0 | ManxPower, I see... ufffff why this PRI card has NT/TE jumpers ?? arghhhh |
| 17:17.13 | ManxPower | kink0, maybe it's not a PRI card. |
| 17:17.18 | ManxPower | Maybe it's a BRI card. |
| 17:17.34 | DarkFlibble | has the timebomb bug been fixed? |
| 17:17.56 | ManxPower | The only PRI cards that are compatable with asterisk that I know of are Digium, Sangoma, and Voicetronix |
| 17:18.16 | ManxPower | And if you are using BRI then I cannot help you in any way at all. |
| 17:18.42 | jpk | www.junghanns.net |
| 17:18.42 | kink0 | ManxPower, this PRI card is not in the PC, is in the 2N Gateway |
| 17:18.54 | jpk | single and double PRI cards and quadro/octo BRI cards. |
| 17:18.54 | caio1982 | brad_mssw, asteriskmonkey: as far as i know a patch is just need in the case of dtmftotext module with 1.2, the rx/tx fax stuff builds and loads fine for me too, probably it's a asteriskmonkey bug :) |
| 17:19.02 | kink0 | and I have connected 2N Gateway PRI to Digium PRI card |
| 17:19.19 | tzanger | kink0: did you flip the card to E1 mode for the span you're using? |
| 17:19.26 | ManxPower | kink0, if it asks about TE or NT mode then it is NOT a PRI card. |
| 17:19.29 | warmcat | DarkFlibble: it is gone for me in 1.2.3 |
| 17:19.36 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-198.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:19.36 | kink0 | tzanger, yes , was on E1 mode, that was right |
| 17:19.43 | DarkFlibble | k... |
| 17:19.44 | *** join/#asterisk goldy2 (i=goldenol@114.sub-70-209-31.myvzw.com) |
| 17:19.45 | goldy2 | hi |
| 17:19.51 | kink0 | tzanger, I get OK status yet, but calls are refused now. |
| 17:19.52 | goldy2 | anyone used the Thomson ST2030 phone yet? |
| 17:19.53 | elvisthedj | Question about a DIY MWI: I'm using an old school classic black deskphone connected to a Supura FXS and just wondering if anybody knows of any plans to add a MWI to something like this (I know it might be specific to the FXS, but I thought there might be some standardized voltage sent down the line to blink the light or something.. not having much luck with google) I've seen these old school phones w/ lights wired up in hotels.. just wondering if an |
| 17:19.53 | elvisthedj | ybody had a thought |
| 17:19.53 | tzanger | ok, now did you plug in a loopback cable? does the span go green within 30 seconds? |
| 17:19.55 | ManxPower | tzanger, He's using a card in a gateway, not Asterisk |
| 17:20.14 | goldy2 | The phone sounds like I am under water --- wondering if there are any Asterisk specific tweaks, voip-info didn't seem to have any. |
| 17:20.15 | tzanger | ManxPower: He was hooking a TE405 up to a gateway I thought |
| 17:20.41 | ManxPower | tzanger, the gateway wants ot know if the the card needs to be in NT mode or TE mode. |
| 17:21.12 | kink0 | tzanger, ManxPower I had spoken about BOTH cards, while green on 2N card, was RED alert on Digium... but I found a configuration where is both green |
| 17:21.32 | kink0 | and call is sent to the 2N, but I got "refuse" at SIP |
| 17:21.57 | kink0 | I then move from TE to NT the 2N card, and from cpe to net, and viceversa the digium card.. |
| 17:22.20 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 17:22.26 | kink0 | but only way all appears ok, and asterisk report it as ok, is TE on the 2N, and net on the digium. |
| 17:22.48 | *** join/#asterisk roulduke_ (i=p40xxdjp@p508D2164.dip0.t-ipconnect.de) |
| 17:23.17 | kink0 | I am setting so again... at least appears be ok, but two errors: claim not found D channel, and use 16 as ussual... and SIP calls get a refuse SIP message. |
| 17:25.17 | tzanger | kink0: you will need asterisk in pri_net mode, since the gateway is expecting to talk to a telephone switch |
| 17:25.28 | tzanger | and then you need to get the signaling right. I am going to guess you want euroisdn or q.sig |
| 17:26.07 | rajiv|work | elvisthedj: it doesn't work for you right now ? i have an analog phone connected to a tdm400p and the light comes on whenever there is the stutter dialtone |
| 17:26.33 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 17:27.07 | asteriskmonkey | caio1982: i just cant get it to load, id doesnt say why |
| 17:27.09 | elvisthedj | rajiv_, There is no light.. I need to add one to the phone. Trying to find a schematic. I might just have to either find one that's been modded, or start probing around for voltage |
| 17:27.27 | asteriskmonkey | caio1982: should i have removed my old one before i redid it? |
| 17:27.46 | rajiv|work | can someone explain why/how http://bugs.digium.com/file_download.php?file_id=9099&type=bug fixes the 1.2.2 issues ? and how it is that the issue only comes up aftr a certain date/time ? |
| 17:28.28 | *** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com) |
| 17:28.28 | |vinsik| | anybody heard of problems with linksys repeater function.. i use 5 WAP54G's.. and it seem to kill datatransfer that tries to the same host and port. |
| 17:29.15 | rajiv|work | elvisthedj: oh. cant' help you there. sorry. but aren't there external boxes you can plug in ? |
| 17:30.01 | kink0 | tzanger, yes, now appears is working, no RED alert or so |
| 17:30.08 | kink0 | tzanger, but I got : ot SIP response 603 "Declined" |
| 17:30.09 | Peste | vinsik: 5 repeaters? |
| 17:30.10 | elvisthedj | rajiv_, Where's the fun in that :D I'm just ready to go dremel crazy on this phone. I might have found some info on sandman.com |
| 17:30.42 | kink0 | while I try to send a call from a SIP to the Asterisk , and asterisk uses the Zap/g1 to carrie the call to the 2N gateway |
| 17:31.18 | brad_mssw | rajiv|work: well, I would assume the difference in ms between ast_tvnow() and 0 overflows a signed integer, so it becomes a negative number |
| 17:31.20 | |vinsik| | peste: yeah.. one thats Access Point.. all other are wireless repeaters connecting to the one access point creating a network. |
| 17:31.43 | *** join/#asterisk Abbas (i=Abbas@203.81.203.207) |
| 17:31.53 | brad_mssw | rajiv|work: so it was just a coincidence that it happened today .. but there would be multiple points in time that it would happen, or not happen |
| 17:32.04 | saftsack | howto remove a modem in hylafax |
| 17:32.25 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 17:32.34 | |vinsik| | oh and by the way.. in what variable does asterisk save caller extension? |
| 17:32.52 | |vinsik| | voip-info.org is sooo slow from my point of origin. ;) |
| 17:32.59 | |vinsik| | too lazy to check it |
| 17:33.10 | brad_mssw | |vinsik|: heh, it's slow from anywhere, the server is overloaded |
| 17:33.19 | caio1982 | asteriskmonkey: of course, but anyway if you really do know what you're doing, that should not be a big deal |
| 17:33.33 | caio1982 | asteriskmonkey: i just rebuilt here right now and loaded the modules gracefully |
| 17:34.05 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:34.23 | |vinsik| | can the SPI firewall affect network in such a way? |
| 17:34.33 | |vinsik| | in linksys u cant disable it. :( |
| 17:34.48 | *** join/#asterisk [1]EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 17:35.11 | elvisthedj | speaking of being too lazy to check the wiki.. just wondering if I'm going to be forced into AEL anytime soon?? |
| 17:35.24 | bkw__ | haha |
| 17:35.27 | bkw__ | AEL |
| 17:35.31 | bkw__ | thats funny |
| 17:35.39 | elvisthedj | i'll take that as a no :D |
| 17:35.44 | |vinsik| | :D |
| 17:36.00 | bkw__ | oh don't get me started |
| 17:36.33 | Err | |vinsik|: stateful packet inspection can only hurt if the state times out - which isn't very likely to happen in a real-time bitstream :-) |
| 17:36.48 | Err | well, unless it decides that the state is "bad" - but typically SPI ignored UDP, other than a timeout |
| 17:36.59 | *** join/#asterisk ibob63 (n=hp@82.111.121.101) |
| 17:37.01 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 17:37.07 | brad_mssw | elvisthedj: well, considering asterisk converts ael into an extensions.conf format internally, no ... |
| 17:38.57 | elvisthedj | brad_mssw, ohhhh.. well then to hell with it. I just thought i'd grab cvs one day and get some "Sorry, extensions.conf is deprecated" message or something. I had to rewrite a bunch of crap because DBget and put are gone from cvshead now. |
| 17:38.58 | *** join/#asterisk tuxinator_linux (n=tuxinato@m410e36d0.tmodns.net) |
| 17:39.21 | bkw__ | makes ya wonder just how much testing goes on if the 1.2.2 thing crept in |
| 17:39.46 | elvisthedj | sorry for whining since I'm a home user :) I suppose it's the people with 10k line dial plans that would be in a really bad mood |
| 17:40.19 | brad_mssw | bkw__: well, that bug wouldn't be seen until the overflow caused the signed integer to become negative, which didn't happen between the timeframe of the commit, and 00:18 or so last night |
| 17:40.26 | *** join/#asterisk dusty (n=Dusty@64.89.118.139) |
| 17:40.48 | bkw__ | brad_mssw, still why was it broken in the first place |
| 17:41.04 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:41.16 | *** join/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net) |
| 17:41.17 | bkw__ | cruise control and regression testing on commit would be helpful |
| 17:41.21 | brad_mssw | bkw__: they were fixing another bug in relation ot call time limits being inaccurate |
| 17:41.26 | mogorman | obviously bkw |
| 17:41.28 | mogorman | go write some |
| 17:41.28 | Reverend | eeek my * stopped sending audio to anyone |
| 17:41.33 | mogorman | oej has started |
| 17:41.36 | tainted_ | dude that fucking sucked |
| 17:41.38 | Reverend | can someone look at my debug log and help me out please |
| 17:41.38 | brad_mssw | bkw__: the actual patch that caused the bug is here : http://svn.digium.com/view/asterisk/trunk/channel.c?r1=7801&r2=7826 |
| 17:41.47 | Err | bkw__: that's a great statement, but regression testing won't catch a bug that only happens in the future |
| 17:42.10 | DarkFlibble | what caused the timebomb? was it malicious? |
| 17:42.12 | bkw__ | hahahahaha I know what they are doing |
| 17:42.17 | SPoon_TSX | Hi, anyone from Ontario Canada here? |
| 17:42.17 | elvisthedj | CLI> flux_capacitor reload |
| 17:42.28 | bkw__ | they are trying to get around the VAD stuff right? |
| 17:42.31 | brad_mssw | DarkFlibble: no, it was not malicious |
| 17:42.33 | lancey | anyone using NuFone? |
| 17:42.36 | DarkFlibble | k |
| 17:42.42 | bkw__ | hacking it in... in the most improper way |
| 17:42.45 | brad_mssw | DarkFlibble: it was due to a signed integer overflow |
| 17:42.48 | bkw__ | yep |
| 17:42.50 | bkw__ | hahah |
| 17:42.54 | mogorman | brian seriously, why do you come here? |
| 17:43.02 | mogorman | you want a plug for freeswitch? |
| 17:43.04 | mogorman | is that what it is |
| 17:43.13 | docelm0 | sigh.. be nice.. |
| 17:43.33 | mogorman | i mean if you want brian |
| 17:43.36 | mogorman | ill change the topic |
| 17:43.38 | mogorman | and you can just go |
| 17:43.43 | bkw__ | mogorman, why do you come to #freeswitch then? |
| 17:43.47 | lancey | anyone using nufone and having no audio :) |
| 17:43.51 | bkw__ | I'm just pointing out the obvious |
| 17:43.53 | mogorman | do i troll in freeswitch? |
| 17:43.58 | bkw__ | I'm not trolling |
| 17:44.00 | mogorman | i like to hear about everything |
| 17:44.02 | mogorman | yes you are |
| 17:44.06 | bkw__ | no i'm not |
| 17:44.10 | *** join/#asterisk Trazz (n=traderz@ip-66-80-141-13.chi.megapath.net) |
| 17:44.13 | brad_mssw | lancey: heh, maybe nufone hasn't upgraded to 1.2.3 yet ;) |
| 17:44.14 | docelm0 | sigh.. would you 2 take this to #bitch |
| 17:44.33 | mogorman | heh docelm0 |
| 17:44.42 | lancey | brad_mssw maybe... but it's nufone, it couldn't be i think, that's why i'm asking |
| 17:44.47 | *** join/#asterisk wzl (n=wzlwzl@wsip-70-183-60-181.oc.oc.cox.net) |
| 17:44.48 | bkw__ | Having half a mind and a real clue about whats going on gets me called a troll |
| 17:44.48 | bkw__ | nice |
| 17:44.53 | bkw__ | oh well |
| 17:44.54 | bkw__ | moving on |
| 17:44.59 | mogorman | lol half a clue |
| 17:45.00 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 17:45.04 | mogorman | yeah id say you have that |
| 17:45.11 | bkw__ | I didn't say half a clue |
| 17:45.14 | bkw__ | learn to read |
| 17:45.20 | rob0 | boys! |
| 17:45.22 | elvisthedj | wow... #Springer all the sudden |
| 17:45.25 | docelm0 | or ladies.. |
| 17:45.30 | bkw__ | haha |
| 17:45.33 | docelm0 | depends on how you look at it.. |
| 17:45.39 | mogorman | heh you see boondocks elvisthedj ? |
| 17:45.41 | brad_mssw | bkw__: how do you figure you know what's going on? your statements made no sense to that point |
| 17:45.53 | bkw__ | brad_mssw, its ok |
| 17:45.58 | saftsack | hylafax -> modem -> tdm fxs card -> isdn |
| 17:46.01 | saftsack | is this possible? |
| 17:46.02 | elvisthedj | mogorman, nope |
| 17:46.21 | Reverend | DEBUG[2522]: channel.c:644 ast_queue_frame: Dropping voice to exceptionally long queue on IAX2/66.234.228.170:4569-2 |
| 17:46.28 | mogorman | in recent episode they discovered to start a riot all you need to do is throw a chair |
| 17:46.39 | evilrabbi | why would asterisk output everything to a tty by default? |
| 17:46.39 | elvisthedj | fighting in #asterisk is just going to lead to a bunch of threatening milicious prank calls |
| 17:46.41 | *** join/#asterisk SGM (n=stoyan@213.91.216.130) |
| 17:47.00 | *** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 17:47.00 | MasterYoda | where is the fire? |
| 17:47.02 | rob0 | "You can't fight in here! This is the War Room!" |
| 17:47.05 | mogorman | heh MasterYoda in the house |
| 17:47.08 | wzl | trying to debug our setup.. have polycom ip300 phones with several broadvoice lines. inter-office calls sound great. calls to the outside world, have several problems from 1) ~.5 sec delay; 2) clipping; 3) other person not hearing anything, us hearing them fine... can anyone suggest a troubleshooting procedure |
| 17:47.35 | tainted_ | bkw__ anyone know if teliax was affected by the 1.2.2 issue? |
| 17:47.54 | [TK]D-Fender | wzi : confirm your codec choice, latency to broadvoice, general bandwidtch, Server lag if transcoding. |
| 17:48.13 | *** join/#asterisk theunabonger (i=Bhongwan@lunacity.themoon.org) |
| 17:48.34 | bkw__ | tainted I would bet on it |
| 17:48.38 | bkw__ | if they were running it |
| 17:48.43 | *** join/#asterisk pdugas (n=pdugas@h73.90.40.69.ip.alltel.net) |
| 17:48.56 | elvisthedj | omg.. i just found a diy neon lamp kit for six bucks.. i knew there were other geeks in the world besides me (well, and you guys) |
| 17:49.17 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 17:49.20 | elvisthedj | :D |
| 17:49.35 | wzl | [TK]D-Fender: in sip.conf, we're only allowing ulaw... office bw is 1.5M up/down... the problem occurs with no one using the internet, and just 1 person on the phone... sever is dedicated 2.4G/512M, with no significant cpu loads |
| 17:49.38 | cpm | <PROTECTED> |
| 17:49.45 | tainted_ | bkw__ i spent hours yesterday debugging scripts, tcpduming, checking configs, recompiling asterisk before we came upon the conclusion that it was a timebomb |
| 17:50.22 | Reverend | what could cause native bridging to stop working? It's not a codec problem... help plz... also tied to this other error that spams the hell out of my console: DEBUG[2522]: channel.c:644 ast_queue_frame: Dropping voice to exceptionally long queue on IAX2/66.234.228.170:4569-2 |
| 17:50.22 | bkw__ | tainted you just realized this ;) |
| 17:50.26 | cpm | <PROTECTED> |
| 17:50.42 | tainted_ | bkw__ no, this was yesterday |
| 17:50.54 | bkw__ | tainted tiz ok you didn't catch that ;) |
| 17:51.46 | [TK]D-Fender | wzl : Whats your latency to Broadvoice? |
| 17:51.48 | Damin | Anyone care to elaborate on the timebomb? :) |
| 17:51.56 | Reverend | oh shit... missed the topic |
| 17:52.01 | Reverend | hope this update fixes my problem |
| 17:52.14 | elvisthedj | by the way.. it's sad to me that the supura fxs (SPA-1000) i bought for 40 bucks is so much better than my toasted IAXy that was 100 bucks. |
| 17:52.18 | knight_ | where is twisted??? |
| 17:52.37 | wzl | [TK]D-Fender : 123 packets transmitted, 123 received, 0% packet loss rtt min/avg/max/mdev = 13.361/29.718/143.478/16.499 ms, pipe 2 |
| 17:52.49 | [TK]D-Fender | elvisthedj : IAXY .. star of the "not quite ready for prime time" show! |
| 17:52.51 | SPoon_TSX | [TK]D-Fender: May I ask your opinion on if I need around 6 - 8 PSTN connection. Is it I better go for PRI route? Will the PRI also having the same echo issue as my POTS had? |
| 17:53.03 | [TK]D-Fender | wzl : interesting... |
| 17:53.21 | wzl | more annoying than interesting, but yea =) |
| 17:53.34 | [TK]D-Fender | SPoon_TSX : Depends on what a PRI will cost you. Echo is universal and will happen on any PSTN interface. |
| 17:53.45 | [TK]D-Fender | wzi : Talked with Broadvoice about it? |
| 17:54.08 | ibob63 | I am about to install asterix but need to buy a phone. can anyone recommend an ip-phone? is there a particular brand that is recommended? |
| 17:54.16 | elvisthedj | [TK]D-Fender, that's putting it pretty nice. It's the first piece of electronics I've actually fried with static... pretty sure it's dead. Sad that I'll never get to provision it again. I used to love to do that every few days |
| 17:54.18 | SPoon_TSX | [TK]D-Fender: Oic. Have you also have a very bad experience with the echo on TDM Card in the past too? |
| 17:54.19 | wzl | no... figured they'd tell me to buy/use their equipment instead |
| 17:54.29 | fugitivo | ibob63: it's asterisk, not asterix |
| 17:54.31 | theunabonger | The Linksys/Sipura/Cisco SPA 941 is a wicked good phone. |
| 17:54.42 | theunabonger | and cheap too. |
| 17:54.56 | [TK]D-Fender | SPoon_TSX : PRI give you better signalling (disconnect, DID info for special routing, faster dialing/answering, etc). POTS is typically cheaper. Its a question of what a given solutions costs, and is worth to you. |
| 17:54.58 | ibob63 | fugitivo: excuse my dyslexai |
| 17:55.00 | Telamon | SPoon_TSX: PRI gives you lots of nice things like caller ID and DID capabilities that you can't do as easily with POTS lines. PRI will probably be more expensive though. |
| 17:55.18 | [TK]D-Fender | SPoon_TSX : I've been royally pissed at my TDM22B and my 2 TE405P's. |
| 17:55.37 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 17:56.01 | [TK]D-Fender | theunabonger : SPA-941 is pretty decent, but too closely priced to the Polycom IP501 for me to suggest it to anyone except a really cheap home user. |
| 17:56.07 | SPoon_TSX | [TK]D-Fender: So there is no advantage on echo cancellation on PRI.... too bad. |
| 17:56.20 | elvisthedj | wonder if i can send this thing to digium for a refurb or something. i only had it for a month |
| 17:56.44 | [TK]D-Fender | SPoon_TSX : Each can have EC on it.... they are just equally susceptable |
| 17:56.47 | theunabonger | hmm |
| 17:57.14 | [TK]D-Fender | theunabonger : The SPA-921/922 would be a better choice in thta case (once released) |
| 17:57.16 | brookshire | elvisthedj: yes.. you can call support |
| 17:57.25 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:57.46 | enigma___ | blah BRIstuff-0.3.0-PRE-1i not only has b0rked timestamps from the future, junghanns.net also forgot to bump the patched in versionstring vom "h" to "i" for the console |
| 17:57.48 | brookshire | once they determine it is bad, they can issue you an rma |
| 17:57.52 | [TK]D-Fender | Calling support = free, GETTING support... well :) |
| 17:57.58 | SPoon_TSX | [TK]D-Fender: Do you recommend purchasing a Echo Cancellation Card for the echo cancellation? |
| 17:58.15 | elvisthedj | stupid question: if i bought an external mwi, i'd want the FSK one, right? |
| 17:58.37 | Telamon | Does anyone know of a way to diagnose the IAX queuing code? We're getting some errors that seem to lead back to a delay in handling IAX network packets, which is resulting in voice cutouts on IAX calls. SIP calls work fine. |
| 17:58.38 | [TK]D-Fender | SPoon_TSX : Wherever you can get EC in hardware I suggest that route. Confim the specs & perfomance, reputation, and reports of users. |
| 17:58.42 | wzl | [TK]D-Fender: i just made a call into a 3rd party voicemail (outside of the office)... while the autoattendant talks, if i try to talk, "she" gets very choppy, cutting in an out.... |
| 17:58.47 | elvisthedj | brookshire, I didn't buy it from them.. not sure if it was an official reseller or not, so i might just be screwed.. That's why i was going for refurb instead of refund :) |
| 17:58.50 | theunabonger | I've had the Cisco 7960, the Cisco 7912, the (yuch) Grandstream JunkPhone BT101, the WiFi Hitachi 5000, and the Linksys SPA-941, and (except for it's inherent lightness, which causes it to move around when answered, and sometimes fall off the desk), it's the best one of the bunch for functionality, and UI. |
| 17:58.58 | elvisthedj | brookshire, but it won't hurt to ask |
| 17:59.09 | SPoon_TSX | oic |
| 17:59.15 | tainted_ | bkw__ ok i just got it |
| 17:59.17 | bsdfreak | heh |
| 17:59.20 | bsdfreak | sup bkw |
| 17:59.20 | tainted_ | bkw__ i'm slow haha |
| 17:59.25 | wzl | [TK]D-Fender: it's almost like its half duplex, in a way =/ |
| 17:59.31 | bsdfreak | anyone have some recommendations for a good fxo card? |
| 17:59.36 | bsdfreak | only needs to be a single line |
| 17:59.44 | theunabonger | anyone unsing NuFone, did they explode/were taken over by terrorists? |
| 17:59.46 | [TK]D-Fender | theunabonger : It just needs better rubber feet... I agree on that |
| 17:59.51 | mogorman | there is only one bsdfreak x100p |
| 17:59.52 | bsdfreak | the: both |
| 18:00.01 | theunabonger | [TK]D-Fender, yup, for sure. |
| 18:00.38 | theunabonger | all your NuFone service is belong to us! |
| 18:00.51 | DarkFlibble | theunabonger, they just haven't patched/upgraded yet |
| 18:01.01 | [TK]D-Fender | you have no chance to transcode! Make your codec! |
| 18:01.03 | DarkFlibble | big bug day! |
| 18:01.14 | *** join/#asterisk lahaine (n=lahaine@167.68.119-80.rev.gaoland.net) |
| 18:01.44 | theunabonger | no bugs for us "slackers", |
| 18:01.57 | theunabonger | I'm using version 1.0.8 ;-P |
| 18:01.57 | MrChimpy | it's a pretty classy bug anyway. not many manage to break systems all over the world at the same time. |
| 18:02.07 | [TK]D-Fender | SPoon_TSX : How many ports are you looking for again? |
| 18:02.10 | theunabonger | oh yeah, what did I miss? |
| 18:02.17 | theunabonger | got a pointer? |
| 18:02.27 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:02.28 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:02.32 | generalhan | whats up guys ! |
| 18:03.11 | DarkFlibble | 12 machines to patch... |
| 18:03.34 | DarkFlibble | the pain... |
| 18:03.37 | DarkFlibble | :P |
| 18:03.44 | generalhan | So ive been having all these problems with my T-1 and i finally figured out what has been wrong. my provider says they dont offer a PRI, and what they have is known as a "2 way PBX Trunk" can i still adjust * to work that way or do i need to find a provider that can offer a PRI ??? |
| 18:03.51 | *** join/#asterisk lorinc (n=ang@caracas-3816.adsl.interware.hu) |
| 18:03.52 | theunabonger | a band? Sodding off? Crikey, it must be GMT! |
| 18:04.03 | theunabonger | have fun MrChimpy ! |
| 18:04.20 | theunabonger | Us N.A. West Coast guys are just staring to work. |
| 18:04.23 | theunabonger | starting |
| 18:04.31 | *** part/#asterisk pdugas (n=pdugas@h73.90.40.69.ip.alltel.net) |
| 18:04.35 | DarkFlibble | MrChimpy, thats 3 people in the uk then... |
| 18:04.38 | generalhan | lol right ... i just got here an hour ago ! |
| 18:04.56 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 18:04.57 | MrChimpy | aye. i've been to your freaky time zone :) |
| 18:04.59 | theunabonger | damn first thing I find is "no incoming lines working" |
| 18:05.07 | theunabonger | heh, |
| 18:05.12 | theunabonger | yeah, we're freaky and we love it. |
| 18:05.18 | generalhan | its true ! |
| 18:05.24 | generalhan | good 'ole West Coast ! |
| 18:05.25 | Zodiacal | anyone know why my work can't receive phone calls from cell phones? verizon says its not their fault. i get the error lady saying "were sorry, your call can not be completed as dialed, if you feel you have reached this recording in error, please check the # and try again"... |
| 18:05.32 | bkw__ | ok i'm back |
| 18:05.36 | Zodiacal | i have tried tmoble and cingular |
| 18:05.40 | Zodiacal | they both get the same message |
| 18:05.43 | theunabonger | Zodiacal, who is your VOIP provider? |
| 18:05.51 | Zodiacal | theunabonger no voip, just verizon |
| 18:05.55 | Zodiacal | pots |
| 18:06.00 | Zodiacal | with a hunt |
| 18:06.06 | theunabonger | probably an SS7 problem. |
| 18:06.15 | theunabonger | er, perhaps. . . . |
| 18:06.19 | Zodiacal | they said they tried reseting the hunt |
| 18:06.22 | Zodiacal | whats ss7? |
| 18:06.30 | Zodiacal | what can i tell them to try next, they don't think its their fault |
| 18:06.39 | DarkFlibble | switching system 7... the protocol the pstn runs |
| 18:06.47 | _Sam-- | are you sure it works fine when you call from a land line |
| 18:06.59 | Reverend | ok, patch fixed no audio :-p |
| 18:07.01 | Zodiacal | i have a few numbers here i have tried |
| 18:07.05 | Zodiacal | and they all work |
| 18:07.10 | Reverend | should have known better than to use the latest version |
| 18:07.17 | Zodiacal | but when calling any of the 8 lines, from cell phones i get that message |
| 18:07.52 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-62bc6e5b08725735) |
| 18:07.54 | Zodiacal | darkflibble can i tell them to reset the ss7 or somthin :) |
| 18:07.57 | Zodiacal | im at a loss |
| 18:08.13 | DarkFlibble | Zodiacal, not easily... |
| 18:08.20 | _Sam-- | in order to call a verizon pots number you have to use a verizon wireless phone |
| 18:08.20 | hhoffman | hi, anyone have a example of creating dynamic meetme conferences from a authenticated user? |
| 18:08.24 | DarkFlibble | you could ask them to troubleshoot it... |
| 18:08.25 | _Sam-- | <just kidding> |
| 18:08.30 | DarkFlibble | but they probably wont |
| 18:08.39 | Zodiacal | sam hehe, the tech had a verizon cell and it worked, you might not be far off |
| 18:08.52 | Zodiacal | darkflibble they came out and said its not their fault.. |
| 18:08.58 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:09.10 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 18:09.12 | _Sam-- | maybe some kind of interconnect point is down someplace |
| 18:09.26 | jpablo | Hi, any horror histories in the asterisk bug? |
| 18:09.50 | Zodiacal | sam-- interconnect point? like out side the building? |
| 18:09.50 | generalhan | So ive been having all these problems with my T-1 and i finally figured out what has been wrong. my provider says they dont offer a PRI, and what they have is known as a "2 way PBX Trunk" can i still adjust * to work that way or do i need to find a provider that can offer a PRI ??? |
| 18:10.29 | _Sam-- | hmm i have 2 way PBX trunks over IAX |
| 18:10.34 | _Sam-- | works fine with * |
| 18:10.47 | generalhan | hmm |
| 18:10.57 | generalhan | well what do i need to change settings-wise to make that work : |
| 18:10.58 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 18:11.03 | _Sam-- | that is a really general statement -- many things could be considered 2 way pbx trunks. |
| 18:11.05 | Damin | FUCK.. |
| 18:11.08 | theunabonger | phone companies suck, which is why we all use asterisk, yes? |
| 18:11.08 | Zodiacal | sam any final suggestions i can try? |
| 18:11.15 | Damin | I think the latest update to 1.2 broke g729.. |
| 18:11.15 | generalhan | do you mind doing a pastebin on your zaptel.conf and zapata.conf for me so i can see ? |
| 18:11.15 | Zodiacal | theunabonger YES |
| 18:11.36 | fugitivo | generalhan: you should ask them what kind of signalling is that |
| 18:11.37 | DarkFlibble | YUP... |
| 18:11.40 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 18:11.41 | _Sam-- | my two way pbx trunks dont use a zap interface |
| 18:11.43 | Damin | I just updated, and now depite having disallow=all and allow=g729, we are returning no compatible codecs. |
| 18:11.58 | Damin | g729 isn't even listed in the codec list for the peer.. |
| 18:12.01 | generalhan | ohh well maybe thats my issue is that im trying to use ZAP |
| 18:12.02 | DarkFlibble | :P |
| 18:12.19 | DarkFlibble | I speak... they listen... not arguing |
| 18:12.27 | fugitivo | generalhan: you need to know the signalling type to setup the zap interface |
| 18:12.32 | fugitivo | and channels |
| 18:12.38 | _Sam-- | generalhan : as fugitivo smartly suggested, you need to find out what signal/ connection type your box is receiving |
| 18:12.43 | generalhan | they sent a tech out here yesterday and he was having them test the lines .. and the lasy on the other end was reading off my servers MAC address so i know that we can see eachother through the T-1 but i just cant make any calls |
| 18:13.32 | _Sam-- | i dont have experience setting up the digium cards for a data t1, so i cant give you any settings there |
| 18:13.46 | generalhan | "data T-1" ? |
| 18:14.04 | theunabonger | heh, all T1's are data. |
| 18:14.09 | theunabonger | and some of that data is voice. |
| 18:14.12 | generalhan | our Data T-1 is fine ... that is not connected even to the same network as our Voice -T-1 ? |
| 18:14.32 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:14.50 | generalhan | if i could just get the signalling correct then id be ok, but these retards are looking at me like i have 5 heads when i talk about signalling and timming and such |
| 18:15.00 | harryvv | generalhan is this your first experiance working with asterisk? |
| 18:15.10 | generalhan | neg |
| 18:15.14 | harryvv | k |
| 18:15.24 | generalhan | i have had a working system up for over 6 months now ... but VoIP and IAX2 not ZAP |
| 18:15.31 | generalhan | this IS my first attempt at a ZAP config |
| 18:15.48 | harryvv | zap is just ordinary pstn |
| 18:16.02 | *** join/#asterisk astnoob_ (n=astnoob@static-209-137-161-171.t1.cavtel.net) |
| 18:16.17 | generalhan | see thats my issue i know NOTHING about PSTN, and Trunks, i went to school for data netowrks not voice |
| 18:16.21 | harryvv | you have a uniqe situation that ive never seen |
| 18:16.45 | harryvv | I think this is a signaling issue |
| 18:16.47 | _Sam-- | which, where the customer doesnt know what kind of circuit they have? |
| 18:16.55 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 18:16.56 | Hmmhesays | pakistani's are freaking hard to deal with |
| 18:16.59 | generalhan | well i called them up and talked to a tier 3 tech and asked him about Switchtype, framing, coding, and timming and he was acting like i was making these terms up |
| 18:17.32 | _Sam-- | maybe you have DSL! |
| 18:17.39 | generalhan | "switchtype" i dont even know what youre talking about, you must want the Data Services tech support |
| 18:17.40 | harryvv | generalhan, or was that your impression? |
| 18:17.43 | fugitivo | generalhan: time to find another provider then... |
| 18:17.47 | generalhan | HAHAHAHA "maybe i have DSL" haha |
| 18:18.01 | _Sam-- | just kidding, obviously :) |
| 18:18.05 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 18:18.09 | tainted_ | docelm0 hey did u get a chance to send the box out? |
| 18:18.10 | generalhan | lol |
| 18:18.12 | jhiver | hi all |
| 18:18.17 | tzanger | hah |
| 18:18.20 | *** part/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 18:18.22 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 18:18.32 | harryvv | generalhan are you doing this for a company? |
| 18:18.57 | jhiver | Is there an easy way for a program to tell asterisk to set / unset a variable which can then be used in the dialplan? |
| 18:19.00 | _Sam-- | i mean, if its a digital data circuit, it can only be two things, a PRI or a T1...and anyone who put it in would know which signal type is needed, at least in my opinion |
| 18:19.05 | jhiver | or do you have to use an AGI? |
| 18:19.07 | `lyme | anyone in here happen to be good with mssql? |
| 18:19.10 | generalhan | well i just got off the phone with them again and it seems that they DO offer a PRI, but now i get to wait a week and a half for another tech to get out here to take a look at our system. apparently they cant just switch me over to a PRI from what i have from their HQ (again i could have called them out on that caues it seems as if they should be able to do it all from their end ... but i know nothing) ! |
| 18:19.12 | tzanger | that's like the bell tech I talked to.. "2915112 is a regular business line at <faraway location>, it says that right here!" "So how the hell am I taking 30000 minutes a month on that number on my PRI then?" "I don't know how your system is set up, sir, but it's clearly a business line over at <faraway address>" |
| 18:19.16 | harryvv | jhiver yea php |
| 18:19.22 | fugitivo | generalhan: where are you located? |
| 18:19.29 | generalhan | AZ |
| 18:19.34 | generalhan | phoenix area |
| 18:19.40 | jhiver | ideally I would like Asterisk to do <this> is some file exists, <that> otherwise |
| 18:19.48 | FuriousGeorge | can someone tell me whats wrong with my attempt at using svn: svn checkout http://svn.digium.com/svn/libpri/trunk/ /usr/src/ |
| 18:19.57 | fugitivo | jhiver: gotoif |
| 18:19.59 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 18:20.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:20.13 | oogle | ok, fess up... who's responsible for the no-sound-bridge bug? |
| 18:20.22 | file | it's all my fault... yup |
| 18:20.25 | file | er wait |
| 18:20.26 | file | no it isn't! |
| 18:20.28 | jhiver | ok I'll take a look at gotoif command |
| 18:20.29 | generalhan | lol |
| 18:20.32 | tzanger | oogle: I'm kind of curious about that too, if only to just tease |
| 18:20.34 | oogle | to late! you said it's your fault |
| 18:20.40 | harryvv | jhiver, you want to make a graphical user interface that will input lines into the config files and delete them? |
| 18:20.43 | wunderkin | ya, file runs eshizzlefonizzletele |
| 18:20.44 | file | bah fine |
| 18:20.50 | jhiver | no |
| 18:21.03 | jhiver | I want to touch a file via ssh when my screen saver starts |
| 18:21.07 | tainted_ | what is svn exactly |
| 18:21.12 | jhiver | and remove this file when my screen saver stops |
| 18:21.13 | file | subversion |
| 18:21.19 | *** join/#asterisk _ZeroOne (n=ZeroOne@81.189.57.6) |
| 18:21.21 | tainted_ | ok |
| 18:21.23 | wunderkin | FuriousGeorge, download instructions are on the asterisk.org download page |
| 18:21.25 | tainted_ | now what is subversion |
| 18:21.26 | _ZeroOne | hello |
| 18:21.26 | jhiver | and I want asterisk to direct to voicemail if the file exists |
| 18:21.28 | harryvv | no comment |
| 18:21.28 | hhoffman | subversion... it's a version control system similar to CVS |
| 18:21.31 | Hmmhesays | ok, ser or openser if i want to do some radius auth |
| 18:21.32 | fugitivo | i knew i shouldn't upgrade to 1.2.2... |
| 18:21.37 | jhiver | so I'm not woken up by the phone when I'm sleeping |
| 18:21.40 | fugitivo | jhiver: gotoif |
| 18:21.43 | jhiver | ok |
| 18:21.50 | harryvv | jhiver, thats not what you want to do |
| 18:21.56 | _Sam-- | why be bleeding edge (1.2.2) in a production environment anyway |
| 18:21.56 | oogle | tainted-: allows multiple people to work on a bunch of source code at once, and keep track of revision... go to their site for more info |
| 18:21.58 | tainted_ | who was affected by 1.2.2 |
| 18:22.01 | harryvv | jhiver, ever try asterisk at home? |
| 18:22.07 | tainted_ | oogle thanks! |
| 18:22.07 | jhiver | harryvv, why? |
| 18:22.09 | jhiver | explain |
| 18:22.15 | fugitivo | i'll not upgrade to 1.2.3 neither "yet" |
| 18:22.44 | oogle | _Sam--: when 1.0.x isn't good enough, trunk isn't stable enough, and there are bugs in 1.2.1 |
| 18:22.55 | jhiver | harryvv, why wouldn't I want to do that? |
| 18:23.14 | tainted_ | oogle so true |
| 18:23.17 | harryvv | its a version of asterisk that you can program you can direct incomming calls to voice mail at a cirtain time of day. say when a bussiness closes, it will automaticly goto a ivr that says the bussiness is closed and goto voice mail |
| 18:23.27 | asteriskmonkey | fugitivo: they just fixed the bridge bug this morningn in 1.23 |
| 18:23.39 | fugitivo | asteriskmonkey: i know |
| 18:23.39 | jhiver | harryvv, that is not what I want |
| 18:23.42 | _Sam-- | i am not asking this to be a smart ass...but whats wrong with 1.2.1 for you? |
| 18:23.55 | jhiver | I sleep next to my desk you see |
| 18:23.57 | harryvv | so you dont want it to ring when you are asleap |
| 18:24.00 | tainted_ | _Sam-- i think 1.2.1 had the same issue |
| 18:24.01 | asteriskmonkey | is there a change list between 1.2 and 1.23 posted yet? |
| 18:24.05 | fugitivo | asteriskmonkey: but i'll keep 1.2.1 |
| 18:24.08 | tainted_ | _Sam-- i had to rollback to 1.2.0 |
| 18:24.09 | jhiver | and don't want to be woken up when I'm asleep, business hours or not |
| 18:24.18 | chiardon | If sometimes, I use to have some extensions that don't work suuddenly and became idle withiut reason . . .whre I can resoleve this bad issue? |
| 18:24.21 | jhiver | so the screensaver makes more sense to me |
| 18:24.31 | harryvv | linux screen saver? |
| 18:24.36 | jhiver | yeah |
| 18:24.49 | jhiver | using xscreensaver-command -watch I can detect events |
| 18:25.01 | FuriousGeorge | wunderkin: now that you meantion it, besides not specifying a local flie nothing is wrong with my syntax. when a do specify a local file i still get the same error |
| 18:25.01 | FuriousGeorge | svn checkout http://svn.digium.com/svn/libpri/trunk libpri-trunk |
| 18:25.02 | FuriousGeorge | svn: REPORT request failed on '/svn/libpri/!svn/vcc/default' |
| 18:25.02 | FuriousGeorge | svn: REPORT of '/svn/libpri/!svn/vcc/default': 400 Bad Request (http://svn.digium.com) |
| 18:25.07 | harryvv | Ask one of the developers here |
| 18:25.08 | Nugget | I used to use that to clear my ssh-agent keys. |
| 18:25.14 | *** part/#asterisk bkw__ (n=brian@70.103.248.130) |
| 18:25.14 | Nugget | quite simple to work with |
| 18:25.57 | jhiver | is there a way for an asterisk condition to evaluate a shell command by any chance? |
| 18:25.58 | *** join/#asterisk Trazz (n=traderz@ip-66-80-141-13.chi.megapath.net) |
| 18:26.00 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 18:26.09 | Nugget | jhiver: sure, write an agi app |
| 18:26.14 | jhiver | daaah ok |
| 18:26.17 | _Sam-- | jhiver: you oculd do what you want in like 3 lines in php |
| 18:26.24 | jhiver | yeah I know |
| 18:26.43 | jhiver | I was looking for an even lazier way of doing it :) |
| 18:27.19 | _Sam-- | netgeeks can probably do it from a dialpan instead of AGI |
| 18:27.47 | *** part/#asterisk enota_ (i=dimka@freelsd.net) |
| 18:28.27 | tzanger | does anyone remember that little win32 utility that tied in with asterisk and popped up caller ID on incoming calls? |
| 18:28.30 | jhiver | OK |
| 18:28.41 | jhiver | Can you set a persistent asterisk variable from the AGI? |
| 18:29.00 | jhiver | so I could do it remotely via ssh using asterisk -rx '<command>' |
| 18:29.02 | DarkFlibble | tzanger, there was a plugin for jabber that did that... |
| 18:29.05 | DarkFlibble | only one I know |
| 18:29.06 | jhiver | sorry |
| 18:29.14 | jhiver | s/AGI/CLI/ |
| 18:29.15 | tzanger | DarkFlibble: yes but there was a standalone util too that looked pretty cool... bah |
| 18:29.25 | tzanger | DarkFlibble: I wrote the jabber plugin :-) |
| 18:29.34 | _Sam-- | how would the standalone w32 app get the caller id, of say, a call coming into your hardware phone |
| 18:29.38 | tzanger | or at least ONE of them. :-) |
| 18:30.02 | jhiver | I seem to remember that asterisk has some database registry like mechanism |
| 18:30.05 | jhiver | could I use that? |
| 18:30.13 | jhiver | aha |
| 18:30.18 | jhiver | looks like I can :) |
| 18:30.18 | tzanger | jhiver: Set(DB/key) or something? |
| 18:30.22 | jhiver | yep |
| 18:30.27 | tzanger | I've never used that yet :-) |
| 18:30.29 | jhiver | that would make more sense |
| 18:30.49 | *** part/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 18:30.55 | jhiver | I could be doing like DBput("DndIsTrue", 1) |
| 18:31.17 | tzanger | yeah something like that |
| 18:31.18 | caio1982 | i suppose it's faster to use the internal DB than any other solution to store and read quickly simple variables that are volatile |
| 18:31.34 | jhiver | ooh that'll be nice |
| 18:31.40 | fugitivo | mmm |
| 18:31.41 | caio1982 | DBput wasn't deprecated anyway? |
| 18:31.42 | jhiver | then I have about ZERO programming to do :) |
| 18:31.48 | Mark_Halverson | last night I had SVN loaded on FC4 = 1.2.2 was working fine....now there is a definate NAT issue...no audio either direction when behind NAT....before when having NAT issues I would atleast get one way audio |
| 18:31.53 | jhiver | I can do it with shell, ssh and that's it :) |
| 18:31.55 | fugitivo | why you need to use the internal db for volatile variables? |
| 18:32.23 | caio1982 | fugitivo: sorry, i didnt meant "volatile" really |
| 18:32.27 | fugitivo | just use variables |
| 18:32.37 | caio1982 | explained above :) |
| 18:32.40 | fugitivo | oh, ok :) |
| 18:33.08 | caio1982 | sometimes you just dont need to send/read anything from realtime or flat file and the internal DB is pretty okay for me |
| 18:33.13 | jpablo | Mark_Halverson: check www.asterisk.org |
| 18:33.32 | jhiver | yeah this looks like a nice feature |
| 18:33.33 | caio1982 | although it's berkley database backend, right? i dont know if it's really better to use all the time... |
| 18:33.39 | Mark_Halverson | i read about todays bug....but it works with a public IP |
| 18:33.43 | Mark_Halverson | that's what's strange |
| 18:33.46 | Mark_Halverson | its all NAT |
| 18:33.53 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 18:33.57 | Mark_Halverson | SVN doesn't do nat worth a damn |
| 18:33.58 | *** part/#asterisk astnoob_ (n=astnoob@static-209-137-161-171.t1.cavtel.net) |
| 18:34.00 | Mark_Halverson | lol |
| 18:34.45 | *** part/#asterisk ibob63 (n=hp@82.111.121.101) |
| 18:35.21 | FuriousGeorge | Mark_Halverson: what bug? im trying to use svn for the first time, im behind nat and im getting 400s |
| 18:35.38 | drumkilla | what does subversion have to do with NAT? |
| 18:35.39 | drumkilla | :) |
| 18:35.54 | FuriousGeorge | i dunno, thats why im asking |
| 18:36.03 | DarkFlibble | just moved asterisk onto an old server and got an answer phone message from 2004 |
| 18:36.10 | fugitivo | that's like saying that your irc client doesn't work because you're behind nat |
| 18:36.19 | drumkilla | lol |
| 18:37.11 | Damin | So is Nufone having problems? |
| 18:37.14 | Mark_Halverson | 1.2.2 was working on a hosted box fine....my ip phone is behind nat....after installing SVN....my phone behind NAT has NO audio....customers on public IPs are working fine |
| 18:37.27 | FuriousGeorge | like i said, all i know is that i copied the command from asterisk.org and im still getting "BAD REQUEST" so when Mark_Halverson says theres a nat related bug you cant blame me for wondering |
| 18:37.47 | jhiver | so you guys think this could work? |
| 18:37.47 | *** join/#asterisk kpettit (n=keith@user-0cet2n7.cable.mindspring.com) |
| 18:37.48 | jhiver | exten => s,1,Setvar(isDnd=DBGet(jhiver/isDnd)) |
| 18:37.59 | fugitivo | FuriousGeorge: the bug is related asterisk 1.2.2, not svn |
| 18:38.16 | FuriousGeorge | fugitivo: yeah i realize what hes saying now |
| 18:38.30 | Mark_Halverson | understand...i was under that impress...i'm just trying a workaround for svn |
| 18:39.12 | FuriousGeorge | so wtf is up with my svn checkout? can someone else try this command and tell me if it works for them |
| 18:39.13 | FuriousGeorge | svn checkout http://svn.digium.com/svn/libpri/trunk libpri |
| 18:40.09 | backblue | Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 18:40.09 | backblue | why do i have allways this? |
| 18:41.58 | FuriousGeorge | backblue: u speak german? |
| 18:42.38 | backblue | hoo, ztdummy, forget i figure it out |
| 18:42.50 | Peste | hello again :) |
| 18:42.53 | backblue | FuriousGeorge: no, i should? |
| 18:43.21 | Peste | somebody know what does "Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)" mean? |
| 18:43.55 | FuriousGeorge | backblue: no, i just googled something about that error but it was in german which i dont speek either |
| 18:44.26 | Peste | i speak german :) |
| 18:44.38 | FuriousGeorge | Peste: he fixed it though |
| 18:44.42 | *** join/#asterisk Prival (n=someone@gw-artistours.dsl.primus.ca) |
| 18:44.43 | Peste | ok |
| 18:45.27 | Prival | If there a wat to avoid agents to be forwarded queue call if they have DND activated? I'm using AddQueueMember for login queue members. |
| 18:45.40 | jhiver | oh boody hell that is annoying |
| 18:45.50 | jhiver | I cannot invoke DBput from the command line |
| 18:46.09 | jhiver | sucks :) |
| 18:46.17 | fugitivo | jhiver: what command line? |
| 18:46.25 | jhiver | the CLI |
| 18:46.29 | iCEBrkr | dbput is deprecated |
| 18:46.45 | jhiver | and what will replace it? |
| 18:46.55 | jhiver | sounds like a pretty useful thing to have to me |
| 18:47.05 | iCEBrkr | jhiver: It's been converted to a DB() function |
| 18:47.08 | fugitivo | from the CLI |
| 18:47.12 | fugitivo | you should use database |
| 18:47.24 | jhiver | fugitivo, how so? |
| 18:47.24 | *** join/#asterisk sevrdseot (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 18:47.41 | fugitivo | database <tab> |
| 18:48.04 | jhiver | ok cheers |
| 18:48.42 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 18:49.11 | Peste | soo..... nobody can give me a clue about this error? |
| 18:49.20 | Prival | Anyone ? Is there a way to avoid agents to be forwarded queue call if they have DND activated? I'm using AddQueueMember for login queue members. |
| 18:49.27 | jhiver | DBput DBget is not deprecated in 1 |
| 18:49.33 | jhiver | 1.0.9 is it? |
| 18:49.40 | fugitivo | jhiver: you want to use it from the CLI or dialplan? |
| 18:49.41 | iCEBrkr | jhiver: No |
| 18:49.47 | Cinen | I know it is off topic but does anyone know if there is a tool that will check mysql 4 databases and make changes so it can be upgraded to 5. I am trying to build an Asterisk cluster. |
| 18:49.55 | jhiver | fugitivo, in the dialplan I do this: |
| 18:50.05 | jhiver | [jhiver-incoming] |
| 18:50.06 | jhiver | exten => s,1,Setvar(isDnd=DBget(jhiver/isDnd)) |
| 18:50.06 | jhiver | exten => s,2,Gotoif(${isDnd}?100:200) |
| 18:50.23 | jhiver | and to test it, I set the flag from the CLI |
| 18:50.31 | jhiver | but it doesn't seem to work unfortunately |
| 18:50.38 | jhiver | I must have made a mistake somewhere |
| 18:51.29 | SPoon_TSX | exit |
| 18:52.02 | *** join/#asterisk dsaas (n=root@190.48.217.201) |
| 18:52.07 | drumkilla | jhiver: instead of DBget, you use the following ... |
| 18:52.23 | dsaas | what a dayyyyyyyyyyyyyy |
| 18:52.25 | drumkilla | exten => s,1,Set(isDnd=${DB(jhiver/isDnd)}) |
| 18:52.43 | jhiver | drumkilla, you think this'll work on my 1.0.9 version of *? |
| 18:52.51 | drumkilla | jhiver: oh. no. |
| 18:52.53 | drumkilla | :) |
| 18:53.10 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 18:53.13 | drumkilla | that would be ... exten => s,1,DBGet(isDND=jhiver/isDnd) |
| 18:53.15 | drumkilla | I ... think |
| 18:53.18 | drumkilla | it has been a while, heh |
| 18:53.19 | Katty | really now. |
| 18:53.25 | Katty | a likely story. |
| 18:53.26 | jhiver | drumkilla, thanks :) |
| 18:53.53 | drumkilla | to double check, look at 'show application DBGet' |
| 18:55.01 | burton | hi, is in 1.2.2 issue audio ? |
| 18:55.11 | mishehu | burton: see the toppic |
| 18:55.19 | mishehu | I woke up this morning to that same thing |
| 18:55.23 | jhiver | drumkilla, looks like it works :) |
| 18:55.26 | jhiver | awesome :) |
| 18:55.58 | iCEBrkr | is in 1.2.2 issue audio? |
| 18:56.09 | iCEBrkr | burton: Try using a better translater |
| 18:56.09 | justinu | engrish? |
| 18:56.13 | justinu | lol |
| 18:56.26 | burton | ;) |
| 18:56.30 | iCEBrkr | justinu: Maybe it's yoda speak? |
| 18:57.08 | burton | what is wrong in 1.2.2 ? |
| 18:57.22 | iCEBrkr | That's an odd question. |
| 18:57.27 | iCEBrkr | burton: Are you having problems or not? |
| 18:57.34 | kpettit | http://bugs.digium.com/view.php?id=6349 |
| 18:57.42 | burton | yes |
| 18:57.51 | burton | with sip to iax |
| 18:57.52 | }btorch{ | is anyon ehere using cdr_mysql ? |
| 18:57.52 | kpettit | that's the timebomb description and also how to get around the bug until you can upgrade |
| 18:58.00 | iCEBrkr | }btorch{: I am. |
| 18:58.13 | justinu | wow, that's a lame bug |
| 18:58.14 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 18:58.34 | kpettit | sounds like something you'd expect with IE |
| 18:58.47 | iCEBrkr | haha |
| 18:58.50 | iCEBrkr | justinu: I have to agree |
| 18:58.51 | iCEBrkr | Like WTF? |
| 18:58.58 | }btorch{ | iCEBrkr: I just compiled the module and setup my cdr_mysql.conf file ,, the module also loads fine but it won't connect |
| 18:59.20 | justinu | that's much worse than the rfc2833 bug I found in ast 1.2 |
| 18:59.43 | }btorch{ | iCEBrkr: I have created the asterickcdrdb and asterisk databases and also set the username to asteriskuser with password amp109 |
| 18:59.46 | iCEBrkr | }btorch{: Permissions correct on MySQL's user table? |
| 18:59.48 | _Sam-- | }btorch{ : check your sql error logs |
| 19:00.06 | }btorch{ | iCEBrkr: yeah I can login fine to the DBs on CLI |
| 19:00.10 | *** join/#asterisk Assid (n=assid@59.183.38.248) |
| 19:00.12 | wasim | }btorch{: wrong password |
| 19:00.17 | iCEBrkr | }btorch{: ...why are you telling me your username/password? :) |
| 19:00.20 | Assid | umm.. can someone help me with text sending.. |
| 19:00.36 | iCEBrkr | }btorch{: Could still be a hostname issue |
| 19:00.41 | }btorch{ | iCEBrkr: hehe is not like u gonna crack my system ;-) |
| 19:00.57 | *** join/#asterisk Tjief (i=Tjief@r0x0r.dk) |
| 19:00.58 | iCEBrkr | }btorch{: From the CLI into MySQL it'll look like 'localhost' from an application it could be your IP or hostname. |
| 19:01.06 | }btorch{ | iCEBrkr: its localhost access ... |
| 19:01.16 | iCEBrkr | }btorch{: Well, >>>I<<< wouldn't do it, but there's over 300 people in here... |
| 19:01.22 | Assid | i get this error: Jan 25 13:58:59 WARNING[1361]: chan_sip.c:7249 receive_message: Received message to <sip:301@asteriskhost.ext> from Satish Alwani<sip:satish@asteriskhost.ext>;tag=93059e30, dropped it... |
| 19:01.32 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 19:01.57 | }btorch{ | iCEBrkr: its a standalone non production server so its ok |
| 19:02.01 | iCEBrkr | :) |
| 19:02.04 | iCEBrkr | I'm just saying |
| 19:02.23 | Assid | is there a reason for it to drop it? |
| 19:02.36 | }btorch{ | iCEBrkr: let me check the permission for the hostname and ip |
| 19:02.54 | jaike | is it possible to set up asterisk so calls made on SIP phones where agents are logged in with AgentCallbackLogin, will be an agent channel instead of a SIP channel? |
| 19:03.25 | Assid | but when i call. it works fine |
| 19:03.56 | *** part/#asterisk hhoffman (n=hhoffman@tor/session/x-62bc6e5b08725735) |
| 19:04.19 | Prival | Anyone ? Is there a way to avoid agents to be forwarded queue call if they have DND activated? I'm using AddQueueMember for login queue members. |
| 19:09.22 | *** join/#asterisk techy (n=tecky@66.9.96.115) |
| 19:11.53 | Assid | anyone here using messaging on sip? |
| 19:13.22 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 19:14.58 | svenna_ | hi all |
| 19:15.05 | svenna_ | i ve got a sip prob now |
| 19:15.20 | tainted_ | how is 1.2.3 holding up? |
| 19:15.23 | svenna_ | i can call out - but calls in dont even apear in the cli |
| 19:15.47 | svenna_ | is it posible, that is has to do with dyn ips? |
| 19:17.08 | tainted_ | svenna_ what version? |
| 19:17.19 | svenna_ | 1.2 |
| 19:18.07 | svenna_ | ahem: Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q |
| 19:18.17 | file | Assid: messaging is only supported when you're in a call, and the device uses the same callid as the call for sending the messages |
| 19:18.58 | svenna_ | first i installed 1.2 - but when i installed the bri stuff - it appears, that it loaded Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q be automatic |
| 19:20.10 | *** join/#asterisk NewSole (n=dave@MTL-HSE-ppp175292.qc.sympatico.ca) |
| 19:20.53 | *** join/#asterisk Atacomm (n=dan@ip67-90-98-195.z98-90-67.customer.algx.net) |
| 19:21.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:21.10 | enigma___ | junghanns.net just fixed the unbumped versionstring in BRIstuff 0.3.0-PRE-1i a few minutes ago ;) |
| 19:21.17 | enigma___ | but it still builds asterisk 1.2.2 |
| 19:21.23 | Atacomm | anyone find queues not working in 1.2.3? |
| 19:21.26 | enigma___ | guess a 1.2.3 version will be out shortly |
| 19:22.29 | Atacomm | we upgraded to 1.2.3 today to fix the critical glitch.......and now people are stuck in queues forever and the phones on our end never ring |
| 19:23.34 | denon | see anything on the console? |
| 19:24.03 | }btorch{ | how can I turn on debug on asterisk ? I did -dcvvvvv but stil the same as -cvvvvv |
| 19:24.10 | Atacomm | it says executing Queue, and then it says it turns on hold music, and then nothing |
| 19:24.31 | _Sam-- | nice |
| 19:24.47 | svenna_ | so, do you think, that i cant be called, is due to the junghanns.net version? |
| 19:25.11 | _Sam-- | Atacomm: can i get a dealer app? :) |
| 19:25.15 | _Sam-- | i would call, but.... :) |
| 19:25.39 | file | ha |
| 19:25.42 | file | that was great _Sam-- |
| 19:25.56 | enigma___ | svenna_: nah, i guess ist because you have some mistake in your config, most probably |
| 19:25.56 | _Sam-- | poor guy, i know the feeling |
| 19:26.13 | _Sam-- | my queues work fine in 1.2.1 |
| 19:26.16 | Assid | file: im on call with myself |
| 19:26.17 | _Sam-- | that is all i know |
| 19:26.18 | Atacomm | well, if they could just fix a critical bug without introducing another........ *sigh* |
| 19:26.18 | Assid | and it dont work |
| 19:26.24 | fugitivo | i'm with 1.2.1 and i won't upgrade |
| 19:26.27 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 19:26.28 | svenna_ | ok, thats my hope, because i can work on it ;-) |
| 19:26.54 | denon | Atacomm: dunno, I havent messed with queues on 1.2.3 yet .. I'd suggest bumping anything mission-critical back to 1.2.1 though |
| 19:26.55 | *** join/#asterisk kilobit2001 (n=locid@206-248-158-52.dsl.teksavvy.com) |
| 19:27.07 | svenna_ | but how could i do something wrong? isnt ist just register the channel and thats it? |
| 19:27.08 | denon | (if you're seeing that kind of problem, anyway) |
| 19:27.20 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 19:27.24 | enigma___ | svenna_: i got a Connected to Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i currently running on asterisk (pid = 13000) |
| 19:27.25 | Atacomm | denon: i was running 1.2.0 but it also borked from that problem last night, so we had to upgrade to 1.2.3 to get bridging to work |
| 19:27.27 | enigma___ | and it works fine |
| 19:27.29 | svenna_ | it looks like that to me in the docs |
| 19:27.46 | _Sam-- | hmm and people were complaining that grandstream releases "beta" firmware in their phones....seems like asterisk no diff. |
| 19:27.57 | _Sam-- | using the public for the beta testing |
| 19:28.08 | denon | _Sam--: that's not true. |
| 19:28.11 | enigma___ | (0.3.0-pre1i bristuff contains 2005-01-25 silence fix) |
| 19:28.19 | Atacomm | our queues are using SIP/ as the members |
| 19:28.22 | _Sam-- | i know, im just saying it to stir shit up |
| 19:28.28 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:28.30 | _Sam-- | but thats how it feels sometimes |
| 19:28.43 | svenna_ | enigma___ when did you install it? |
| 19:28.48 | denon | _Sam--: it is, however, important to test any product before you use it in mission-critical production, whether it be a phone system or a floor mat |
| 19:29.22 | _Sam-- | denon: are you a big denon fan? i have this really old, portable, denon DAT recorder :) |
| 19:29.29 | enigma___ | svenna_: a few hours ago, junghanns.net just rereleased the pre1i because they forgot to bump up the version from "h" to "i" |
| 19:29.40 | denon | _Sam--: yeah, they make some ok stuff .. a denon ADAT? |
| 19:29.42 | Atacomm | denon: that is true......until the tested one falls flat on its face from a major date-related glitch, and the only patch kills everything else |
| 19:29.53 | denon | Atacomm: were you having issues with 1.2.1? |
| 19:29.54 | _Sam-- | nope, not adat....just regular DAT...dtr90p |
| 19:29.59 | _Sam-- | i was a big live taper |
| 19:30.04 | svenna_ | ah ok, i go and get it - and pray |
| 19:30.06 | Atacomm | denon: as far as i know we were on 1.2.0 |
| 19:30.17 | denon | Atacomm: might be worth rolling to 1.2.1 for the time being |
| 19:30.32 | denon | I havent been tracking the issues very closely, but we havent had any complaints internally with our 1.2.1 installs |
| 19:31.10 | *** join/#asterisk RoyK (n=roy@ti211310a080-5285.bb.online.no) |
| 19:31.13 | svenna_ | enigma___ didt you configure something special to get it connected? or did you just register the channel? |
| 19:31.27 | denon | Atacomm: may also want to file a bug report (after you check if one has been filed already :) |
| 19:31.37 | JonR800 | heh.. so 1.2.4 should be out soon... |
| 19:32.24 | nroej | hey is there a way to get iax2 running via tcp? |
| 19:32.52 | _Sam-- | denon: do you think there is a way to write data backups to an audio DAT player if it has spdif/optical? |
| 19:32.58 | enigma___ | svenna_: huh? i use it with 2 BRI cards to be able to call and receive calls from anywhere (using iax2) |
| 19:33.06 | denon | _Sam--: eh, sure .. but how fast .. |
| 19:33.10 | denon | I mean, you could always do audio coupling |
| 19:33.19 | _Sam-- | true (how fast) |
| 19:33.26 | denon | worked on tape decks |
| 19:33.46 | }btorch{ | this sux debug isn't helping much on mysql and asterisk |
| 19:33.54 | reza | does anyone else use NuFone? their system has been down/busy all morning |
| 19:33.59 | enigma___ | nroej: you really want voice data via a tcp connection? |
| 19:34.13 | enigma___ | nroej: udp serves that purpose much better |
| 19:34.19 | _Sam-- | reza: i think someone else does use nufone because i saw someone else complaining in here earlier |
| 19:34.20 | }btorch{ | I keep getting this crap .... Failed to connect database server on . Check debug for more info MySQL database sock file not specified. Using default |
| 19:34.36 | reza | i've got a question - if i get 1 DID number, can I only receive 1 call at a time, or unlimited? |
| 19:34.38 | svenna_ | enigma___ ok, i see - (im an asterisk starter) so you didnt configure it in sip.conf, you use iax.conf?!? |
| 19:34.51 | _Sam-- | reza: depends how many simultaneous connections you setup with your provider. |
| 19:35.03 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:35.10 | enigma___ | svenna_: since i use iax2 softphones (beside my regular isdn phones) aye ;) |
| 19:35.15 | reza | what's a good did provider -- i need like 3 simultaneous lines for 1 phone number |
| 19:35.35 | _Sam-- | reza: its all a matter of personal opinion...a good provider is someone that has good routes from where you are |
| 19:35.51 | enigma___ | svenna_: but sip/iax2 doesnt make a that big difference while configuring |
| 19:35.57 | reza | hmm, i need a 925 area code. (northern calif). any thoughts? |
| 19:36.27 | }btorch{ | does * work with mysql 5 ? |
| 19:36.34 | _Sam-- | }btorch{: yes |
| 19:36.56 | saftsack | some hylafax experts here? |
| 19:37.02 | svenna_ | enigma___ ok, so you connected your iax2 phones to asterisk and call out via normal phone line (like isdn or something)? |
| 19:37.11 | _Sam-- | i use cdr_mysql w/ mysql 5 fine |
| 19:37.22 | svenna_ | or do you also call out via iax2? |
| 19:37.31 | _Sam-- | saftsack: you're STILL not faxing?? |
| 19:37.37 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 19:37.39 | _Sam-- | crap its been weeks since we talked about hylafax |
| 19:37.43 | }btorch{ | for some reason my wont' connect |
| 19:38.04 | saftsack | _Sam--, yes because i throwed txfax / rxfax away |
| 19:38.19 | saftsack | and i began 3 hours ago to fax with a simple faxmodem |
| 19:38.23 | enigma___ | svenna_: aye |
| 19:38.23 | saftsack | but it doesnt work :( |
| 19:38.30 | enigma___ | svenna_: that is exactly what i do |
| 19:38.52 | diLLec | saftsack: did you use a modem emulated from ISDN |
| 19:39.01 | enigma___ | i call out via isdn |
| 19:39.03 | _Sam-- | saftsack: probably doesnt work because you didnt get a modem from the list |
| 19:39.03 | saftsack | i did use iaxmodem |
| 19:39.11 | diLLec | ah - ok |
| 19:39.11 | enigma___ | but use iax2 to connect to the box |
| 19:39.13 | saftsack | _Sam--, i have a modem from the list |
| 19:39.22 | _Sam-- | which modem |
| 19:39.22 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:39.27 | reza | can you recieve a fax over iax2? |
| 19:39.29 | nroej | enigma___: yes i want to try some nasty things |
| 19:39.40 | reza | (or send) |
| 19:39.47 | nroej | enigma___: running iax via tor which is tcp only |
| 19:40.04 | nroej | but i guess the latency would be too high |
| 19:40.14 | tainted_ | ~seen Darwin |
| 19:40.41 | jbot | darwin <n=darwin2@84-121-76-246.onocable.ono.com> was last seen on IRC in channel #elive, 10d 25m 24s ago, saying: 'thanathermesis vive de elive ?'. |
| 19:40.41 | svenna_ | enigma___ ok! that also works here! my prob is, that i also call to the outside via sip, but when i call the sip-number from the outside it doesnt even show in my cli - i cant reach it from the outside... |
| 19:40.41 | tainted_ | !seen Darwin |
| 19:40.41 | _Sam-- | darwin_35 |
| 19:40.41 | tainted_ | oh |
| 19:41.03 | RoyK | hey ho |
| 19:41.08 | tainted_ | _Sam-- do u work for teliax? |
| 19:41.14 | RoyK | that bug today was rather nasty |
| 19:41.22 | tainted_ | RoyK wasn't it though |
| 19:41.26 | _Sam-- | nope but darwin_35 ford |
| 19:41.29 | _Sam-- | er ford = does |
| 19:41.30 | saftsack | _Sam--, no i am building hylafax from source |
| 19:41.37 | saftsack | is there any make samples function? |
| 19:41.49 | [TK]D-Fender | ~seen jbalcomb |
| 19:41.51 | jbot | jbalcomb is currently on #asterisk (1d 3h 13m 26s). Has said a total of 57 messages. Is idling for 22h 14m 52s, last said: 'by his mom'. |
| 19:41.51 | saftsack | or just faxsetup? |
| 19:41.58 | _Sam-- | if you make, make install, faxaddmodem, and faxsetup |
| 19:42.00 | _Sam-- | you should be good |
| 19:42.05 | _Sam-- | you forgot faxaddmode |
| 19:42.06 | _Sam-- | m |
| 19:42.11 | reza | Does * surpport T.38 fax relay? |
| 19:42.11 | tainted_ | any issues with 1.2.3 so far? |
| 19:42.14 | redder86 | what the heck is the "timebomb" bug? * channel.c: Issue #6439 - the "timebomb" bug. I'm not seeing 6439 on bugs.digium.com |
| 19:42.15 | saftsack | i didnt forgot faxaddmodem |
| 19:42.23 | saftsack | i could read beforce modem is ok and idle |
| 19:42.25 | Katty | redder86: i should consult with you |
| 19:42.32 | reza | red - last night * died |
| 19:42.34 | JonR800 | tainted_: possibly issues with queues |
| 19:42.40 | redder86 | reza: no doubt |
| 19:42.40 | reza | if you changed your clock back, it would work again |
| 19:42.40 | Katty | redder86: another Important Matter of color combinations. |
| 19:42.47 | _Sam-- | how are you submitting the faxes to hylafax? |
| 19:42.47 | tainted_ | JonR800 what issues are u having |
| 19:42.50 | reza | it died at a specific type |
| 19:42.59 | redder86 | Katty: just as soon as I get freakin' 1.2.2 blown away |
| 19:43.01 | JonR800 | tainted_: Atacomm was having them, customers stuck in queue. |
| 19:43.04 | tainted_ | reza how's it going? what version did u land on |
| 19:43.05 | Katty | redder86: sure thing. |
| 19:43.09 | Katty | redder86: take your time. |
| 19:43.17 | Katty | redder86: i've got all month. ...i think |
| 19:43.24 | reza | the patched svn from last night; any updates? |
| 19:43.28 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 19:43.33 | redder86 | so what's this timebomb bug that rendered my IAX2 communications worthless? |
| 19:43.36 | reza | my provider is down today (nufone) so i can't test :( |
| 19:43.43 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 19:43.46 | johnnyb | Is it possible to disable a ZAP line from a group while asterisk is running? |
| 19:43.56 | _Sam-- | reza: for as cheap as it is, you should (could) get a cheap backup provider |
| 19:43.56 | reza | redder86 - did you read what i said? |
| 19:44.02 | tainted_ | redder86 what went down |
| 19:44.15 | RoyK | i have been wondering for that, johnnyb |
| 19:44.18 | saftsack | _Sam--, what to do with faxmodem? |
| 19:44.25 | RoyK | just disable a PRI |
| 19:44.27 | reza | sam - i'm just getting started with * and it was recommended by a friend. got another one that's reasonably easy to setup and cheap? |
| 19:44.31 | redder86 | reza: I read, but you didn't give me a technical enough answer |
| 19:44.33 | *** join/#asterisk FaH||r (i=Juventus@212-200-201-167.smin.sezampro.yu) |
| 19:44.36 | rt | When I try to modprobe ztdummy, I get FATAL: Error inserting ztdummy (/lib/modules/2.6.14-1.1656_FC4/misc/ztdummy.ko): No such device |
| 19:44.45 | rt | any ideas? (I'm on a FC_4 system) |
| 19:44.49 | johnnyb | I'm using TDM, not PRI |
| 19:44.54 | _Sam-- | reza: your mileage will vary...but i use www.iax.cc as a backup to teliax, and its worked good for ME |
| 19:44.57 | saftsack | _Sam--, i submit them with sendfax |
| 19:45.02 | redder86 | tainted_: SIP seems to be working, but IAX2 doesn't pass audio |
| 19:45.24 | tainted_ | redder86 what version |
| 19:45.30 | redder86 | 1.2.2 of course |
| 19:45.35 | tainted_ | redder86 could be that your provider hasn't updated |
| 19:45.43 | redder86 | tainted_: I am my own provider. |
| 19:45.45 | reza | redder86 - look at the patch for what changed. basecially a bug was introduced 2 weeks ago that manifested itself at a specific time for some reason. i didnt dig into it too much. |
| 19:45.46 | tainted_ | redder86 you should roll back to 1.2.0 or go with 1.2.3 |
| 19:45.59 | iCEBrkr | Is there anything I need to do other than create an inbound context to answer a PRI?? |
| 19:46.06 | _Sam-- | saftsack: when you submit the fax, you dont see anything in your log files? |
| 19:46.13 | *** part/#asterisk FaH||r (i=Juventus@212-200-201-167.smin.sezampro.yu) |
| 19:46.20 | redder86 | reza: still not technical enough |
| 19:46.22 | *** join/#asterisk FaH||r (i=Juventus@212-200-201-167.smin.sezampro.yu) |
| 19:46.27 | iCEBrkr | 4 |
| 19:46.29 | iCEBrkr | errr oops |
| 19:46.34 | iCEBrkr | missed ^A |
| 19:46.35 | redder86 | reza: what kind of bug manifests itself on a time delay? |
| 19:46.39 | reza | redder - how is 'look at source' not technical? |
| 19:46.39 | saftsack | _Sam--, no |
| 19:46.57 | saftsack | request id is 5 (group id 5) for host localhost (1 file) |
| 19:47.02 | saftsack | this is the onliest message i got |
| 19:47.04 | redder86 | reza: give me a URL to the patch so I don't have to look at the diff |
| 19:47.07 | reza | redder did you even look at the patch (do you code?_ |
| 19:47.14 | _Sam-- | do you see the fax in /var/spool/hylafax/ anywere? |
| 19:47.14 | redder86 | reza: I code. |
| 19:47.15 | brad_mssw | redder86: simple, a signed integer overflow |
| 19:47.20 | saftsack | Modem ttyS0 (+49.06XXX.XXXXXX): Running and idle |
| 19:47.24 | reza | http://bugs.digium.com/view.php?id=6349 |
| 19:47.25 | _Sam-- | if it gave you a number (5), then it accepted your fax |
| 19:47.28 | redder86 | brad_mssw: thank you sir |
| 19:47.29 | iCEBrkr | grrrr |
| 19:47.33 | saftsack | _Sam--, do you mean the file? |
| 19:47.40 | iCEBrkr | My number continues to spew "This number is disconnected...." |
| 19:47.43 | reza | so does www.iax.cc work right now? |
| 19:48.06 | _Sam-- | saftsack: in /var/spool/hylafax/sendq or any other directory, do you see your fax or things related to it |
| 19:48.08 | tzanger | reza: avoid iax.cc |
| 19:48.29 | saftsack | my fax isnt in sendq |
| 19:48.30 | brad_mssw | redder86: the bug just wasn't apparent in the timeframe from when the bug was introduced, until last night ... but it would eventually start working again ... and stop working again .... ah, the nature of integer overflows ... especially since they are tracking difference in milliseconds |
| 19:48.56 | reza | tzanger - why? got another recommendation? |
| 19:48.57 | _Sam-- | check the other directories, saftsack |
| 19:49.07 | [TK]D-Fender | iCEBrkr : Shove in some "catch-all" entries to see how the DID is registering.Make sure that if you're expecting 10-digits, that you are getting 10 digits. Jim's uses "4" digits (freakish) |
| 19:49.27 | _Sam-- | saftsack: from what you are telling me so far, it sounds like you just dont have the modem setup right to dial out |
| 19:49.27 | iCEBrkr | [TK]D-Fender: 4 digits isn't that odd really. |
| 19:49.35 | reza | saftsack - what are you trying to do wrt faxing? i need to figure out how to hook up a real fax to * and make it work both ways |
| 19:49.50 | iCEBrkr | [TK]D-Fender: But like, where do I set that up? I'm assuming the inbound call via PRI is a lot like an inbound call via VoIP, no? |
| 19:49.52 | _Sam-- | that is a function of /var/spool/hylafax/etc/dialrules |
| 19:49.54 | _Sam-- | or something thereabouts |
| 19:50.03 | redder86 | brad_mssw: yeah, I understand an integer overflow. |
| 19:50.21 | saftsack | cannot find the file anywhere :( |
| 19:50.32 | *** join/#asterisk vixtor (n=Victor@193.226.149.66) |
| 19:50.35 | saftsack | reza, ? |
| 19:50.37 | redder86 | brad_mssw: all IAX providers using 1.2.1 and 1.2.2 or corresponding CVS versions went down, huh? |
| 19:50.48 | redder86 | brade_mssw: that's horrid |
| 19:50.50 | _Sam-- | saftsack: type "locate dialrules" |
| 19:50.52 | reza | tzanger - have you had bad luck with iax.cc? |
| 19:50.55 | brad_mssw | redder86: no, 1.2.1 was unaffected from what I've heard |
| 19:50.57 | _Sam-- | as long as your updatedb has run recently |
| 19:50.59 | saftsack | i can find dialrules |
| 19:51.00 | jhiver | YES LADS IT WORKS :) |
| 19:51.07 | saftsack | /var/spool/hylafax/etc/dialrules |
| 19:51.18 | jhiver | this is such a cool hack I have no words :) |
| 19:51.19 | brad_mssw | redder86: the bug was introduced in SVN r7826 |
| 19:51.25 | _Sam-- | i have dialrules.europe if you need it |
| 19:51.27 | redder86 | brade_mssw: so if I were using openpbx I would have been spared this, eh? |
| 19:51.30 | reza | jhiver - what are you trying to do? |
| 19:51.32 | tzanger | reza: when it works it works |
| 19:51.34 | saftsack | _Sam--, that would be cool :) |
| 19:51.40 | brad_mssw | redder86: that was Jan 5 ... not sure when 1.2.1 was released |
| 19:51.42 | tzanger | reza: but if it doesn't, good fucking luck getting it from them |
| 19:51.48 | svenna_ | enigma___ im installing 0.3.0-PRE-1i (* 1.2.2) right now... |
| 19:51.49 | vixtor | guys, thank you :-) just by entering here I found out what kept me busy all day long :-) |
| 19:51.52 | vixtor | the no audio problem.. |
| 19:51.59 | jhiver | reza, when my screensaver is on, my extension goes on voicemail directly |
| 19:52.04 | vixtor | i am now downloading 1.2.3 and trying it.. |
| 19:52.18 | vixtor | can you imagine just entering an irc channel and have your problem solved before saying anything? |
| 19:52.27 | *** part/#asterisk FaH||r (i=Juventus@212-200-201-167.smin.sezampro.yu) |
| 19:52.42 | saftsack | _Sam--, behind a router here :( |
| 19:52.55 | redder86 | brad_mssw: good thing I didn't upgrade all of my customers to 1.2.2 :-) |
| 19:52.56 | brad_mssw | heh, isn't everyone behind a router :) |
| 19:53.09 | brad_mssw | maybe you mean NAT or firewall or similar |
| 19:53.10 | _Sam-- | just msg me your email, i am still not positive that this will work without modification, but it is a good start. |
| 19:53.21 | fdask | when i use the dial command, and two lines get bridged, i immediately get a crazy feedback noise |
| 19:53.22 | Atacomm | so far going back to 1.2.1 is giving me all circuits are busy on the PRI.....grr |
| 19:53.31 | fdask | any quick suggestions what might be causing this? |
| 19:53.43 | fdask | i dropped my line gains down to their lowest settings, same issue |
| 19:53.44 | jpablo | Atacomm, did you also downgraded zaptel ? |
| 19:53.44 | saftsack | _Sam--, why are you not sure? are there any good guides for hylafax in germany? |
| 19:53.50 | jaike | fdask: 1.2.2? |
| 19:53.53 | fdask | ya |
| 19:54.00 | [TK]D-Fender | iCEBrkr : Yes, your PRI should have a context. Thats where you should put an "i" and "_x." catch-all to see what's coming in |
| 19:54.07 | *** join/#asterisk areski (n=areski@229.Red-83-44-69.dynamicIP.rima-tde.net) |
| 19:54.07 | _Sam-- | dialrules are just like the extensions.conf for asterisk...you ahve to tell the modem how to dial based on where you are |
| 19:54.08 | Atacomm | jpablo: no, i was just thinking about that..... i forgot i didnt go to 1.2.2....... i had zaptel 1.2.2 |
| 19:54.13 | jaike | fdask: upgrade to 1.2.3 |
| 19:54.13 | _Sam-- | and my dialrules are setup for where i am |
| 19:54.16 | redder86 | _Sam-: the hylafax-users@hylafax.org mailing list is excellent you should try that |
| 19:54.22 | saftsack | _Sam--, ok |
| 19:54.23 | jpablo | Atacomm, anyway, better to upgrade to 1.2.3 |
| 19:54.24 | redder86 | _Sam-: I may even reply :-) |
| 19:54.27 | iCEBrkr | [TK]D-Fender: Thing is, I don't even see the call coming in. |
| 19:54.30 | Atacomm | i just did, it hosed my system |
| 19:54.38 | Atacomm | queues with SIP/ don't work in 1.2.3 |
| 19:54.39 | saftsack | _Sam--, is hylafax hard to configure? |
| 19:54.44 | _Sam-- | redder86: i been on it for years.. :) |
| 19:54.51 | [TK]D-Fender | iCEBrkr : Are you sure the DID is targeting YOUR PRI and that its "active"... I remember how these guys are... |
| 19:54.54 | _Sam-- | saftsack: not in my opinion! |
| 19:54.56 | redder86 | _Sam-: me too |
| 19:55.06 | saftsack | _Sam--, is there any good logger? |
| 19:55.09 | _Sam-- | if you did faxsetup, and faxaddmodem, you are probably close to done |
| 19:55.14 | _Sam-- | mine logs to syslog |
| 19:55.25 | saftsack | syslog shows nothing here |
| 19:55.29 | saftsack | ooh f***** |
| 19:55.32 | saftsack | im working on ssh |
| 19:55.36 | iCEBrkr | [TK]D-Fender: Well, I'm just trying to cover my bases here.. My friend was telling me that if the card isn't setup to answer that line then you're gonna get the disconnected message. Now he's just providing me with general PBX/Telco knowledeg |
| 19:55.39 | saftsack | and my syslog shows me my own pc |
| 19:55.45 | saftsack | thats soo stupid |
| 19:55.52 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 19:55.53 | saftsack | wondering why syslog shows nothing |
| 19:55.56 | reza | jhiver - sounds fucking cool; are there docs for it written up someplace? |
| 19:56.01 | [TK]D-Fender | iCEBrkr : Yeah, if you don't answer they should get the disconnected message |
| 19:56.08 | [TK]D-Fender | maybe a "busy" instead |
| 19:56.12 | _Sam-- | tehre is also stuff in /var/log/hylafax/logs |
| 19:56.30 | jhiver | now... I wonder if voicemail call back would be doable in asterisk? |
| 19:56.34 | iCEBrkr | [TK]D-Fender: I know when my asterisk box is down, VoicePulse gives a "The person you're trying to reach is not accepting calls at this time." |
| 19:56.39 | batphone | im at a loss as to where to start troubleshooting large amounts of dropped calls |
| 19:56.54 | jhiver | i.e. try to ring my extension every couple of hours for as long as I have voicemail or something |
| 19:56.57 | batphone | the console just says the normal stuff about disconnection |
| 19:56.57 | *** part/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 19:57.01 | batphone | nothing unusual |
| 19:57.04 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 19:57.04 | _Sam-- | batphone: its easiest to just blame it on the grandstream gxp2000 phones |
| 19:57.13 | batphone | the provider says the T1 cards are acting up |
| 19:57.19 | Atacomm | okay, got 1.2.1 working.......suppose i'll get around to filing a bug report once i get caught up |
| 19:57.21 | batphone | but have only reset twice in the last 24 hours |
| 19:57.23 | [TK]D-Fender | iCEBrkr : Well thats because they are the telco for you and are "nice"... T1 carriers assume you're doing the job and feel informing YOUR caller is a "bonus" at best |
| 19:57.32 | batphone | which would explain all calls getting dropped at once |
| 19:57.38 | batphone | but not the intermittent nature of the problem |
| 19:57.39 | Katty | batmobile, now with your own personalized voip batphone |
| 19:57.43 | redder86 | okay, 1.2.3 seems to work ... for now at least until the next new thingy overflows |
| 19:57.48 | batphone | like every 5 minutes a call is dropped |
| 19:57.52 | jhiver | I suppose I could have some .call files generated... |
| 19:58.07 | iCEBrkr | [TK]D-Fender: Those to reasons are why I'm trying to determine if the issue is on my end or not. |
| 19:58.09 | jhiver | is there a way in asterisk to check if a given voicemail number has messages in it? |
| 19:58.20 | redder86 | if I hit another thing like this, though, I swear I'm gonna go back to 1.0 |
| 19:59.11 | tuxinator_linux | Katty: at least I thought it was funny, batmobile |
| 19:59.16 | tainted_ | redder86 don't go with bleeding edge |
| 19:59.22 | saftsack | _Sam--, /var/log/hylafax is empty |
| 19:59.27 | iCEBrkr | [TK]D-Fender: I should at least see a message in the CLI if the call is getting to my Sangoma card.... |
| 19:59.30 | _Sam-- | should be |
| 19:59.35 | _Sam-- | <PROTECTED> |
| 19:59.56 | Katty | tuxinator_linux: my stupid humor is wasted in here. |
| 20:00.14 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-50-54.38-151.net24.it) |
| 20:00.18 | redder86 | tainted_: the choices are two: 1.0 ... which is way out of date, 1.2 which bleeds |
| 20:01.12 | redder86 | tainted_: I upgraded my Asterisk box here at home office and figured I should upgrade Asterisk too. |
| 20:01.18 | brad_mssw | actually, 1.2.0 and 1.2.1 were relatively sane |
| 20:01.24 | sevrdseot | so, 1.2.3 is a critical update from 1.2.2, what about 1.2.1 ? buggy as hell? |
| 20:01.38 | brad_mssw | i think the main thing is too much untested stuff is still being put into 1.2 |
| 20:01.44 | brad_mssw | when that should all go into HEAD only |
| 20:01.50 | jhiver | reza, it's very simple |
| 20:01.57 | jhiver | I'll write a page on it |
| 20:01.58 | jhiver | hang on |
| 20:02.09 | saftsack | _Sam--, Jan 25 21:01:18 debian FaxQueuer[12367]: SUBMIT JOB 8 |
| 20:02.15 | saftsack | this are the onliest messages i get |
| 20:02.16 | fugitivo | is any way to change AGENTS monitor files? |
| 20:02.21 | fugitivo | (not queues) |
| 20:02.26 | [TK]D-Fender | iCEBrkr : So put in the 2 catch-alls, NoOp them so you know which one caught it and there's your answer. |
| 20:02.34 | fugitivo | i want to put a custom filename for agents recordings |
| 20:02.50 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 20:02.53 | iCEBrkr | [TK]D-Fender: I still don't see anything even at verbose 9 |
| 20:02.55 | redder86 | Okay, so the only lingering thing in my 1.0 -> 1.2 conversion is my music on hold. |
| 20:03.11 | [TK]D-Fender | iCEBrkr : I'd question if the DID is even targeting your PRI then. |
| 20:03.15 | _Sam-- | saftsack: your jobs are being submitted fine to your faxserver (hylafax) but something is going on when they are being processed for queuing and delivery....you will have to see why. maybe permissions, maybe you dont have all the required programs to convert the fax to the proper outgoing format |
| 20:03.22 | [TK]D-Fender | iCEBrkr : Pastebin your zapata.conf and the targeted context. |
| 20:03.36 | redder86 | Katty: I've got a sec now |
| 20:03.40 | saftsack | _Sam--, ok and howto determine those things? |
| 20:03.51 | redder86 | Katty: but I'm no expert, mind you |
| 20:04.07 | Katty | k |
| 20:04.15 | _Sam-- | probably the best place is hylafax.org and go through the step by step instructions |
| 20:04.21 | redder86 | _Sam-: ah, it's saftsack who's needing HylaFAX help |
| 20:04.23 | reza | what's wrong with my lan?! |
| 20:04.35 | *** join/#asterisk Damin_PDA (n=pocketir@61.sub-70-199-148.myvzw.com) |
| 20:04.46 | _Sam-- | redder86: yeeah, i had pointed him towards hyla a few weeks ago |
| 20:04.50 | saftsack | redder86, hi did you hear from me |
| 20:04.52 | Katty | reza: i ate it |
| 20:04.54 | saftsack | ? |
| 20:05.01 | saftsack | Jan 25 21:04:26 debian FaxSend[12628]: SEND FAILED: JOB 7 DEST 0506102 ERR No local dialton |
| 20:05.04 | saftsack | aaaaaaaaaaaah :) |
| 20:05.19 | _Sam-- | you are making progress |
| 20:05.24 | _Sam-- | it tried to send the fax |
| 20:05.28 | saftsack | theres missing a 0 |
| 20:05.28 | _Sam-- | but as you can see, it says no dialtone |
| 20:05.34 | [av]bani | http://freshmeat.net/projects/asterisk/ |
| 20:05.37 | saftsack | because i am behind a tk |
| 20:05.42 | [av]bani | someone want to update that :P |
| 20:05.46 | *** join/#asterisk oceanlan (n=irc@cpe-71-65-112-38.woh.res.rr.com) |
| 20:06.15 | saftsack | _Sam--, do you know where to add the 0 ? |
| 20:06.21 | _Sam-- | saftsack here is the page that tells you what else you need besides hylafax: (libtiff, ghostscript, etc): http://www.hylafax.org/content/Handbook:Source_Code_Install |
| 20:06.27 | [av]bani | someone should poke mark to update the freshmeat entry |
| 20:06.30 | saftsack | thanks :) |
| 20:06.37 | _Sam-- | saftsack of course i do, to the file called dialplan |
| 20:06.42 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:06.54 | _Sam-- | er dialrules! |
| 20:06.54 | saftsack | thought so too |
| 20:06.55 | _Sam-- | sorry |
| 20:06.56 | saftsack | but how |
| 20:07.09 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:07.12 | _Sam-- | figure it out. |
| 20:07.13 | redder86 | saftsack: http://hylafax.sourceforge.net/howto/install.php |
| 20:07.17 | Seldon1975 | do the 1.2.3 critical updates only apply to PRI users |
| 20:07.18 | _Sam-- | i am down holding hands! |
| 20:07.20 | _Sam-- | er done. |
| 20:07.38 | Katty | so much for emotional support |
| 20:07.42 | _Sam-- | lol |
| 20:07.46 | _Sam-- | i dont have the time |
| 20:08.02 | Seldon1975 | I mean if I'm only using POTS trunk lines I suspect the update isnt that critical...? |
| 20:08.04 | _Sam-- | i have important macaroni salad to eat |
| 20:08.05 | redder86 | saftsack: use the hylafax-users@hylafax.org mailing list ... works like a charm |
| 20:08.12 | Katty | _Sam--: now that's important, that is. |
| 20:08.21 | saftsack | redder86, didnt get this ^^ |
| 20:08.29 | *** topic/#asterisk by twisted[asteria] -> Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) |
| 20:08.30 | saftsack | using a mailing list? do you mean i should simply ask tehre? |
| 20:08.39 | Damin_PDA | katty: I have thouroughly molestedJunky on several occasions for you.. |
| 20:08.44 | Katty | twisted[mobile]: do you provide emotional support? |
| 20:08.48 | Katty | Damin_PDA: thank you dear. |
| 20:08.52 | *** join/#asterisk hans (n=fugalh@hfugal.NMSU.Edu) |
| 20:08.53 | redder86 | saftsack: read the HOWTO here: http://hylafax.sourceforge.net/howto/install.php, then if that doesn't help use the mailing list |
| 20:09.07 | [av]bani | there is POTS but where is PANS |
| 20:09.07 | saftsack | ok thanks |
| 20:09.21 | Katty | [av]bani: _Sam-- needed it for the macaroni salad. |
| 20:09.23 | _Sam-- | saftsack because the macaroni was good...these are the lines i feel you need to mess with in your dialrules file: |
| 20:09.24 | Damin_PDA | twisted: still have my astricon voicemail? |
| 20:09.25 | Seldon1975 | why dont software developer ever write HOWNOTTOs |
| 20:09.27 | _Sam-- | ^${IDPrefix}${Country}${Area} = ! local phone call |
| 20:09.27 | _Sam-- | ^${LDPrefix}${Area} = ! local phone call |
| 20:09.28 | _Sam-- | ^${IDPrefix}${Country} = ${LDPrefix} ! STD call (long distance) |
| 20:09.28 | _Sam-- | ^[+]${Country}${Area} = ! local phone call |
| 20:09.28 | _Sam-- | ^[+]${Country} = ${LDPrefix} ! STD call (long distance) |
| 20:09.28 | _Sam-- | ^[+] = ${IDPrefix} ! international call |
| 20:09.30 | _Sam-- | er sorry all. |
| 20:09.33 | _Sam-- | didnt think it would be that big |
| 20:09.37 | Seldon1975 | aaargh |
| 20:09.37 | twisted[asteria] | Katty, you talking to me or my laptop? |
| 20:09.40 | *** join/#asterisk areski (n=areski@229.Red-83-44-69.dynamicIP.rima-tde.net) |
| 20:09.43 | Katty | twisted[asteria]: you |
| 20:09.53 | [av]bani | Katty: reasonable enough |
| 20:09.57 | Katty | twisted[asteria]: t<tab> |
| 20:10.34 | iCEBrkr | [TK]D-Fender: http://pastebin.ca/38580 |
| 20:10.36 | twisted[asteria] | aww, so i'm first on your tab completion list (or my laptop is) |
| 20:10.51 | diLLec | is there any tutorial about SER and Asterisk availible ? |
| 20:10.52 | twisted[asteria] | and the answer is: when i have time |
| 20:11.50 | De_Mon | i'm using chanspy on a SIP->IAX call through eyeBeam and the audio cuts in and out |
| 20:12.00 | _Sam-- | fyi saftsack, i know people here will say otherwise, but if you are using debian, the hylafax debian package works fine (apt-get install hylafax) |
| 20:12.05 | Katty | twisted[asteria]: i think it's cause you're the last t person to type |
| 20:12.07 | _Sam-- | my last hylafax box was setup that way |
| 20:12.20 | De_Mon | almost like it knows there is silence ahead and stops transmitting too soon |
| 20:12.39 | twisted[asteria] | Katty, hrm. |
| 20:13.00 | saftsack | yes im using this ;) |
| 20:13.08 | Katty | twisted[asteria]: i think i'm going to need emotional support later. |
| 20:13.09 | JonR800 | diLLec: http://openser.org/dokuwiki/doku.php?id=asterisk_integration and http://www.onsip.org/modules/altern8news/ |
| 20:13.16 | De_Mon | _Sam-- can you forward the call from asterisk to hylafax? |
| 20:13.22 | _Sam-- | you said you made from source |
| 20:13.25 | *** join/#asterisk Trazz (n=traderz@ip-66-80-141-13.chi.megapath.net) |
| 20:13.28 | diLLec | cool - thanx JonR800 |
| 20:13.36 | iCEBrkr | CUZ I AIN'T NO HYLAFAX GIRL |
| 20:13.43 | *** join/#asterisk benjk (n=benjamin@70.103.248.130) |
| 20:13.47 | _Sam-- | De_Mon: there are ways to play with asterisk and hylafax together...but mine is all separate...i have seperate pots lines for hylafax |
| 20:13.48 | JonR800 | diLLec: none of which are express start to finish howtos.. but with some reading there's enough info |
| 20:14.06 | diLLec | i see |
| 20:14.07 | De_Mon | _Sam-- ahh |
| 20:14.11 | svenna_ | damn, i installed asterist 1.2.2 over 1.0.10 and now nothing works anymore |
| 20:14.18 | benjk | ~seen zoa |
| 20:14.21 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 3h 35m 48s ago, saying: 'the configs are not completelyt he same'. |
| 20:14.23 | [TK]D-Fender | Few time I been around that fax, now it don't reply to calls and thats a fact.... |
| 20:14.24 | svenna_ | even installing 1.0.10 doesnt work |
| 20:14.25 | iCEBrkr | svenna_: You should read the changelog before doing shit like that.. |
| 20:14.26 | [TK]D-Fender | <PROTECTED> |
| 20:14.32 | iCEBrkr | [TK]D-Fender: lol |
| 20:14.33 | _Sam-- | lol |
| 20:14.33 | [TK]D-Fender | :D |
| 20:15.15 | svenna_ | man, why do i alway touch my running system |
| 20:15.17 | jhiver | OK, I have documented this hack :) |
| 20:15.19 | jhiver | http://ykoz.net/intl/trick/ |
| 20:15.29 | iCEBrkr | svenna_: Cuz you're a geek |
| 20:15.32 | *** join/#asterisk thosa (n=thosa@p54879AE2.dip0.t-ipconnect.de) |
| 20:15.34 | svenna_ | :-) |
| 20:15.35 | iCEBrkr | svenna_: Can't leave things be. :P |
| 20:15.42 | iCEBrkr | svenna_: I know this cuz I'm the same way |
| 20:15.45 | _Sam-- | the ol 'if it aint broke dont fix it' rule |
| 20:15.54 | _Sam-- | it will burn you more often than not |
| 20:15.58 | svenna_ | :) |
| 20:16.21 | jaike | 1.2 features are worth upgrading to |
| 20:16.36 | iCEBrkr | [TK]D-Fender: I'm starting to thing the tech was smoking when he told me there's a number attached to these PRIs |
| 20:16.47 | _Sam-- | they are only worth upgrading to if you need the features |
| 20:16.50 | [TK]D-Fender | iCEBrkr : Couple things wrong with your pastebin : Those aren't patterns, so remove the "_", your "i" doesn't answer first and you should add the NoOp to the 1st line to show the EXTEN. and then add one for "exten => _X,1,NoOp(incoming unknown call on DID *${EXTEN}*) |
| 20:16.52 | iCEBrkr | [TK]D-Fender: I just made a phonecall and the chick said it doesn't appear to have a number associated with it |
| 20:17.13 | [TK]D-Fender | iCEBrkr Could be, but still add in what I suggested |
| 20:17.17 | iCEBrkr | [TK]D-Fender: Yea, I understand the "_" thing, but ya know, it just wouldn't work without on my personal asterisk box.... |
| 20:17.58 | [TK]D-Fender | iCEBrkr : You are mixing bits up..... the Gods of syntax shall enact a toll from you! |
| 20:18.18 | iCEBrkr | lol |
| 20:18.24 | *** join/#asterisk hans (n=fugalh@hfugal.NMSU.Edu) |
| 20:18.34 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 20:19.27 | Telamon | Any IAX wizzards out there? I'm having a problem with voice packets when using IAX2, but SIP calls work fine. |
| 20:19.38 | zoa | Telamon: what is the rtt ? |
| 20:19.39 | iCEBrkr | [TK]D-Fender: Yea, there's no number associated with those PRIs |
| 20:19.41 | zoa | over 150ms ? |
| 20:19.43 | iCEBrkr | Just got confirmation |
| 20:19.57 | [TK]D-Fender | iCEBrkr : f'n brilliant..... |
| 20:19.59 | iCEBrkr | [TK]D-Fender: Dude, I cut-n-pasted all that shit into the pri-in context :P |
| 20:20.01 | Telamon | Getting this in the logs: chan_iax2.c: chan_iax2: ast_sched_runq ran 61 scheduled tasks all at once |
| 20:20.06 | zoa | ouch |
| 20:20.09 | zoa | doesnt sound good |
| 20:20.17 | cpm | [TK]D-Fender, <- is brilliant |
| 20:20.21 | iCEBrkr | [TK]D-Fender: I really don't care if there's a number associated... |
| 20:20.29 | iCEBrkr | [TK]D-Fender: I just figured if there was, I'd use it for our conf line. |
| 20:20.30 | jaike | Telamon: thats an 1.2.0 debug message...that shouldnt be a problem |
| 20:20.36 | Telamon | zoa: How do I find that out? You mean via a ping? This is on a LAN, so it doesn't seem to be a network issue. |
| 20:20.45 | fdask | weird |
| 20:20.55 | fdask | when i bridge calls, i get a really noisy feedback |
| 20:21.02 | fdask | i can change the pitch of it with my voice :V |
| 20:21.10 | zoa | ive its on a lan, that is not the problem |
| 20:21.50 | Telamon | jaike: Hmm, I'm getting voice cutouts every 3-5 seconds that last about 1 second. As I said: SIP works fine, it's just with IAX. I'm really stumped as to what it can be, and that's really the only log message that seems to come close. |
| 20:21.56 | *** join/#asterisk lorinc (n=ang@caracas-1742.adsl.interware.hu) |
| 20:22.14 | iCEBrkr | Hrrm, I wonder how much an 800 number is gonna cost for these PRIs |
| 20:22.18 | *** join/#asterisk arcy (n=arcanum@ppp54-adsl-118.ath.forthnet.gr) |
| 20:22.43 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 20:22.43 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:22.43 | *** part/#asterisk KranZ (n=user@imail.bestline.net) |
| 20:22.45 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 20:22.47 | twisted[asteria] | Oh, Damin, I do have a day job, so it's likely I won't answer your calls when you call at like 2+am |
| 20:22.59 | KranZ | poop |
| 20:22.59 | shmaltz | anybody ever seen asterisk just stopping to bridge calls? |
| 20:23.01 | file | twisted[asteria]: he was probably drunk |
| 20:23.08 | twisted[asteria] | file, so? |
| 20:23.17 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 20:23.17 | file | do you really want to talk to him when he's drunk? |
| 20:23.21 | file | and you're not? |
| 20:23.26 | [TK]D-Fender | iCEBrkr : Typically 800#'s just target a PSTN number which would be a DID targeting your PRI. For which I'm sure the rate would be the same as your outbound from the PRI unless you outsourced the LD on it. |
| 20:23.28 | iCEBrkr | twisted[asteria]: Damin called you at 2am?!?! |
| 20:23.30 | twisted[asteria] | file, i'm usually drunk |
| 20:23.36 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 20:23.43 | iCEBrkr | [TK]D-Fender: I mean for the number, not per min. |
| 20:23.50 | file | twisted[asteria]: wouldn't surprise me |
| 20:23.54 | twisted[asteria] | haha |
| 20:24.10 | [TK]D-Fender | iCEBrkr : All the ones I've worked with they gave me the 800 free... guess they are banking on the minutes for $ |
| 20:24.20 | iCEBrkr | [TK]D-Fender: Oh that'd be nice! |
| 20:24.27 | _Sam-- | schmalts: i think many people have seen that behaviour |
| 20:24.28 | [TK]D-Fender | iCEBrkr : thats the way it is here.... |
| 20:24.32 | *** join/#asterisk razu (n=razu@195.222.10.105) |
| 20:24.51 | iCEBrkr | [TK]D-Fender: Cuz heres the deal.. If this 800# is free and the per min is the same for our outbound, then I'd be saving the company a bunch of money |
| 20:24.53 | _Sam-- | but they are all using 1.2.2or3 or something...you are probably the third or fourth guy today that ive seen make that comment |
| 20:24.55 | [TK]D-Fender | shmaltz : Read the topic |
| 20:25.12 | *** join/#asterisk wspraul (n=wspraul@dbl.q-ag.de) |
| 20:25.17 | [TK]D-Fender | iCEBrkr : Should be... depends on your telco I guess... |
| 20:25.20 | iCEBrkr | yup |
| 20:25.42 | iCEBrkr | [TK]D-Fender: We're paying for a stupid 700 conf number at the moment. $150-$200/mo for conf calls. How lame. |
| 20:25.55 | wspraul | hi there. I had a working chan_capi setup but now switched to a HFC ISDN card for better echo cancellation. |
| 20:26.18 | [TK]D-Fender | iCEBrkr : Yeah, pretty lame. Once I got meet-me running and have our remote agents on eyeBeam too, they all love it. |
| 20:26.20 | shmaltz | [TK]D-Fender thanks, what was the critical problem? |
| 20:26.24 | wspraul | now I'm using bristuff, but it won't work. |
| 20:26.28 | iCEBrkr | [TK]D-Fender: That's the plan |
| 20:26.36 | hans | what if you're running 1.2.1? |
| 20:26.40 | wspraul | Is anybody here who is using a HFC BRI card with bristuff? |
| 20:26.43 | [TK]D-Fender | shmaltz : Bridging stops passing audio as of today :) |
| 20:27.38 | svenna_ | wspraul : jepp, i was -> i just soot it up, but it was working until about 15min ago... |
| 20:27.51 | Err | only for the next 2^31 microseconds :-) |
| 20:28.02 | wspraul | svenl_, great, so you can help me to make my situation worse :-)) |
| 20:28.05 | wspraul | (just kidding of course) |
| 20:28.15 | svenna_ | i see ;-) |
| 20:28.22 | wspraul | Maybe I can help you? |
| 20:28.28 | shmaltz | [TK]D-Fender, but why did it work yesterday? |
| 20:28.32 | wspraul | I do have chan_zap working. |
| 20:28.36 | Aughey | [TK]D-Fender: eyeBeam? |
| 20:28.45 | [TK]D-Fender | shmaltz : The bug kills bridging because of a TIME bug.... |
| 20:28.50 | wspraul | svenl_, if I use chan_zap, then I don't need chan_capi (and capi.conf) anymore, correct? |
| 20:28.52 | [TK]D-Fender | Aughey : Softphone |
| 20:28.59 | twisted[asteria] | svenna_, there's a certain time of day in which it strikes |
| 20:29.02 | svenna_ | i got it all working but updated whithout brain to 1.2.2 |
| 20:29.06 | wspraul | svenna_, if I use chan_zap, then I don't need chan_capi (and capi.conf) anymore, correct? |
| 20:29.19 | Aughey | ahh, xten's softphone |
| 20:29.37 | svenna_ | jepp |
| 20:29.42 | shmaltz | [TK]D-Fender, can you define TIME? |
| 20:29.47 | brad_mssw | Yes, that bug will exist for the next 24.85 days |
| 20:29.52 | brad_mssw | then it will disappear for 24.85 days |
| 20:29.58 | brad_mssw | then reappear for 24.85 days ... and so on |
| 20:30.23 | brad_mssw | assuming that your system handles integer overflows in that manner |
| 20:30.39 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:30.51 | [TK]D-Fender | shmaltz : Follow these guys, and read the bug report |
| 20:31.07 | shmaltz | D-Fender, bug number? |
| 20:31.15 | jaike | brad_mssw: geek stuff |
| 20:31.23 | jaike | hehe |
| 20:31.42 | *** join/#asterisk mroth_imm (n=mroth@63.65.26.220) |
| 20:31.48 | *** join/#asterisk techy_ (n=tecky@66.9.96.115) |
| 20:31.50 | [TK]D-Fender | shmaltz : its in the changelog |
| 20:32.31 | jaike | will wait a full week before upgrading to 1.2.3 ... there might be other bugs lurking |
| 20:34.18 | fugitivo | i'll wait for 1.5 |
| 20:35.20 | *** part/#asterisk mroth_imm (n=mroth@63.65.26.220) |
| 20:35.35 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 20:36.14 | _Sam-- | fugitive by then it will come with a gui ;) |
| 20:36.15 | shmaltz | changelog has a bug in it |
| 20:36.20 | fugitivo | lol |
| 20:36.32 | shmaltz | its not a joke |
| 20:36.43 | fugitivo | lol at _Sam-- |
| 20:37.03 | shmaltz | while the log says bug: http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.3 6439 |
| 20:37.11 | shmaltz | when it actualy is 6349 |
| 20:37.15 | brad_mssw | yep, i mentioned that hours ago :/ |
| 20:37.17 | _Sam-- | fugitivo are you a phone consultant or you do IT work for someone else? |
| 20:37.21 | *** join/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 20:37.43 | fugitivo | _Sam--: consultant |
| 20:38.04 | DynaGuy | hello |
| 20:38.11 | _Sam-- | do you provide hosted PBX services to clients? |
| 20:38.16 | fugitivo | no |
| 20:38.58 | *** part/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 20:39.03 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 20:39.04 | _Sam-- | are you in argentina, or just your host there? |
| 20:39.08 | shmaltz | brad, what function is there that has to do with real date, that made this happen? |
| 20:39.26 | fugitivo | _Sam--: i'm in argentina, why so many questions? :) |
| 20:39.33 | _Sam-- | im just curious! |
| 20:39.51 | fugitivo | you want hosted pbx services in argentina? |
| 20:39.53 | *** join/#asterisk dextro (n=dextro@209.163.224.58) |
| 20:39.53 | wspraul | svenl_, which Asterisk-specific kernel modules do you use? zaptel, zaphfc? others? |
| 20:40.04 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 20:40.04 | wspraul | svenna_, which Asterisk-specific kernel modules do you use? zaptel, zaphfc? others? |
| 20:40.12 | _Sam-- | fugitivo: nah, not yet! hows biz down there? |
| 20:40.18 | VJ | Hi |
| 20:40.20 | wspraul | Do I need qozap? |
| 20:40.32 | fugitivo | _Sam--: well, i'm not going to be rich doing this :) |
| 20:40.43 | _Sam-- | maybe you should move to the USA :) |
| 20:40.46 | VJ | i am facing a problem with asterisk box |
| 20:40.53 | fugitivo | _Sam--: i did, 2 years ago |
| 20:40.57 | _Sam-- | seems there is still plenty "easy money" on the table here |
| 20:41.00 | VJ | i cannot listen what customer says |
| 20:41.05 | fugitivo | _Sam--: but worked for a company |
| 20:41.14 | fugitivo | _Sam--: for 1 year |
| 20:41.15 | SwK[Work] | what time is that thing gonna start hitting up in tehstates on teh audio drops |
| 20:41.25 | Mavvie | VJ: http://www.asterisk.org/blog/73 |
| 20:41.47 | *** join/#asterisk dextro (n=dextro@209.163.224.58) |
| 20:41.56 | Katty | SwK[Work]: hey stranger. |
| 20:42.01 | dextro | any one got bit by the audio bug this morning? |
| 20:42.13 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 20:42.48 | svenna_ | wspraul i use(d?) zaphfc |
| 20:43.13 | wspraul | svenna_, my interrupts are going up like crazy, and I think it's because of timer issues. Also sound quality is lousy. |
| 20:43.20 | svenna_ | right now i try to go back to 1.0.10 to get it working... |
| 20:43.23 | dextro | or know what dependency there was on the date? |
| 20:43.30 | wspraul | svenna_, I am currently looking into rtai (www.rtai.org) |
| 20:43.49 | Katty | SwK[Work]: snob. |
| 20:45.22 | SwK[Work] | sorry |
| 20:45.24 | SwK[Work] | katty |
| 20:45.32 | SwK[Work] | kicking boxen in the head with 1.2.2 on them |
| 20:45.34 | SwK[Work] | heh |
| 20:45.37 | *** join/#asterisk zkillz (i=koma@d145144.adsl.hansenet.de) |
| 20:45.39 | zkillz | hi |
| 20:45.59 | svenna_ | wspraul : i have to tell you, that im a asterisk beginner! |
| 20:46.11 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:46.19 | Katty | SwK[Work]: (= |
| 20:46.29 | *** part/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 20:46.32 | *** join/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 20:47.19 | VJ | Thanx Mavvie |
| 20:47.33 | VJ | i just gone mad, looking into configuration of my system |
| 20:47.38 | *** join/#asterisk sevenn (n=sevenn@205.161.0.4) |
| 20:47.48 | VJ | changed 2 VOIP boxes as well to see whats wrong |
| 20:47.59 | Telamon | VJ: It's probably a codec conversion problem. |
| 20:48.07 | VJ | <PROTECTED> |
| 20:48.18 | VJ | might be |
| 20:48.30 | sevenn | Hi, my name is Cody Jarret and Im having a hard time with my asterisk server. |
| 20:48.31 | dextro | so does anyone know what caused the "timebomb" bug? |
| 20:48.42 | Telamon | VJ: Ah, okay, I used to get that problem when using an IAXy to talk to a SIP phone. I had to disallow gsm for the iaxy, and it started working. |
| 20:48.48 | msw | dextro: the old tick overflow bug? |
| 20:48.50 | sevenn | someone forwarded all of my calls to my mom |
| 20:49.02 | Hmmhesays | so change it back |
| 20:49.25 | svenna_ | wspraul : what is it, you want to do with asterisk? |
| 20:49.29 | wunderkin | it was y2k delayed a few years! |
| 20:49.39 | rob0 | sevenn: your mom did that ;) |
| 20:49.42 | iCEBrkr | lol |
| 20:49.47 | hardwire | anybody have a clue on how snom shared line works? |
| 20:49.53 | wspraul | svenna_, connect ISDN and SIP |
| 20:49.54 | sevenn | no, i think it was cody's mom |
| 20:49.57 | hardwire | and are other phones starting to implement that? |
| 20:50.03 | hardwire | I really enjoy the PTT part of these phones |
| 20:50.08 | [TK]D-Fender | hardwire : "Shared Lines" is not supported in * |
| 20:50.08 | [av]bani | heh |
| 20:50.23 | hardwire | [TK]D-Fender: I didn't ask that :) |
| 20:50.25 | [av]bani | [TK]D-Fender: have you ever used ITSP for DIDs? |
| 20:50.39 | [TK]D-Fender | [av]bani : Not personally but have helped those who do. |
| 20:50.40 | svenna_ | wspraul let me guess, that you r from germany also? |
| 20:50.47 | wspraul | yes |
| 20:51.20 | [av]bani | [TK]D-Fender: have you ever setup one like this: one PSTN @ location call-forwards to ITSP with local DID |
| 20:51.36 | [av]bani | [TK]D-Fender: so that if the ITSP has outage, turn off callforwarding and you can still get some local calls |
| 20:51.41 | _ZeroOne | does a german asterisk irc-channel exist? |
| 20:52.28 | svenna_ | now it does :-) #asterisk.ger |
| 20:52.39 | _ZeroOne | fine :) |
| 20:53.43 | [TK]D-Fender | [av]bani : NOt sure if I got your directions right, but its a little like what I do at home |
| 20:53.46 | wasim | asterisk.de |
| 20:54.03 | Hmmhesays | "you set up our asterisk box 2 year ago, now we're moving ip phones and its not working"\ |
| 20:54.07 | wasim | well, i guess people outside of .de speak gr |
| 20:54.15 | Hmmhesays | ha, that's a shame |
| 20:54.39 | twisted[asteria] | Hmmhesays, heh. |
| 20:54.49 | wunderkin | Hmmhesays, but it worked 2 years ago! |
| 20:55.09 | hardwire | grgrgr |
| 20:55.19 | *** join/#asterisk backblue_ (n=moo@87-196-46-161.net.novis.pt) |
| 20:55.24 | Hmmhesays | no one said a damn thing about moving the phone to a natted environment in the sale |
| 20:55.32 | [TK]D-Fender | [av]bani : My home (analog) forwards to a DID at work if busy/noanswer. Work answers the DID and places the call through SIP to home. So effectively PRI + 1 lines at home. |
| 20:56.46 | *** join/#asterisk johnm_ (n=johnm@10.245.adsl.brightview.com) |
| 20:57.01 | *** join/#asterisk furiousgeorge_ (n=email@pool-70-111-13-159.nwrk.east.verizon.net) |
| 20:57.40 | sevrdseot | quick question: why would the Panel say "connection refused", my webserver is running on port 9050, does that change anything? |
| 20:58.56 | *** join/#asterisk robin_z (n=robin@adsl.redpoint.org.uk) |
| 20:59.28 | *** join/#asterisk FaH||r (i=Juventus@212-200-202-224.smin.sezampro.yu) |
| 20:59.37 | *** part/#asterisk FaH||r (i=Juventus@212-200-202-224.smin.sezampro.yu) |
| 21:01.08 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 21:01.14 | KriS83 | Hi |
| 21:01.45 | VJ | Hello, can anyone suggest me the solution for patch < mydiff.txt |
| 21:01.45 | robin_z | so ... I think I found the cause fo my * crashes ... |
| 21:01.46 | fugitivo | sevrdseot: quick answer, because the connection is refused |
| 21:02.19 | jhiver | darn |
| 21:02.29 | jhiver | can't find out where to place .call files on my xorcom's distro |
| 21:02.31 | sevrdseot | fugitivo: I'm asking why the connection would be refused, why the panel isn't responding |
| 21:02.36 | Mavvie | VJ: that is the solution. |
| 21:02.48 | robin_z | I had numbered the CAPI channel controllers 1,2,3,4 instead of 0,1,2,3 ... under load, when a call was routed to controller 4, I got an illegal controller message, and then it quiteth. |
| 21:02.56 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:03.00 | Seldon1975 | wow - something really odd just happened on our asterisk box: |
| 21:03.01 | Seldon1975 | Zap/20-1 s@incoming:1 Up Bridged Call(Zap/18-1) |
| 21:03.01 | Seldon1975 | Zap/18-1 91@default:1 Up Dial(Zap/g1/1) |
| 21:03.17 | Seldon1975 | two of our trunk lines were tied together |
| 21:03.25 | Seldon1975 | any idea how this could have happened? |
| 21:03.38 | Seldon1975 | even stranger - the apparent destination is '91' |
| 21:03.47 | harryvv | "tied together"? |
| 21:04.02 | Seldon1975 | zap 18 was bridged to Zap 20 |
| 21:04.06 | brad_mssw | uhh, well, what's odd about that ? |
| 21:04.15 | Seldon1975 | both of them are external lines going into the FXS modules |
| 21:04.15 | robin_z | looks "normal" |
| 21:04.15 | brad_mssw | if 18 is FXS and 20 is FXO that's normal |
| 21:04.21 | Seldon1975 | no |
| 21:04.21 | brad_mssw | or call forwarding could do that |
| 21:04.24 | Seldon1975 | they're both FXS |
| 21:04.32 | Seldon1975 | call forwarding |
| 21:04.34 | VJ | Mavvie, i did that |
| 21:04.40 | VJ | but still its not working |
| 21:04.42 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 21:04.44 | brad_mssw | ok, if they're both fxs, then you just have 2 extensions talking to eachother |
| 21:04.59 | VJ | patch updated the channel file of asterisk source |
| 21:05.01 | Mavvie | VJ: after that, make and make install? |
| 21:05.05 | Seldon1975 | i mean they're both Outside lines |
| 21:05.06 | VJ | i did that |
| 21:05.15 | brad_mssw | Seldon1975: ok, they're FXOs then |
| 21:05.15 | Seldon1975 | its like two people from outside talking to eachother |
| 21:05.17 | Hmmhesays | debian oh how I love thee |
| 21:05.18 | Seldon1975 | ok |
| 21:05.19 | Seldon1975 | sorry |
| 21:05.29 | Mavvie | VJ: and restart of asterisk? |
| 21:05.29 | robin_z | so, someone forwarded a call from someoen that dialled in, to an external .. |
| 21:05.30 | Seldon1975 | so that happens with Call Forwarding? |
| 21:05.42 | Seldon1975 | hmm |
| 21:05.58 | brad_mssw | Seldon1975: yeah, if someone forwarded their extension to a cell phone or something, and someone called that extension, it would in turn dial back out which would make sense ... assuming you have call forwarding enabled |
| 21:06.01 | mishehu | hey guys, anybody with polycom soundpoint ip500's know what the configuration setting for the phone to remember the last volume set by the user is in sip.cfg ? |
| 21:06.02 | Seldon1975 | but this is the odd thing: |
| 21:06.11 | Seldon1975 | the destination is '91@default' |
| 21:06.19 | VJ | thanks man |
| 21:06.27 | Seldon1975 | what does that mean? |
| 21:06.33 | robin_z | Hmmhesays: now, if debains so great, why doesnt the MTA work by defualt huh? |
| 21:06.35 | VJ | i did not restarted that |
| 21:06.41 | VJ | now its woring |
| 21:06.45 | VJ | *working |
| 21:06.47 | Mavvie | VJ: heh :-P |
| 21:06.49 | Seldon1975 | no-one would intentionally forward to '91' surely |
| 21:07.01 | Hmmhesays | robin_z: it takes like 6 seconds to setup |
| 21:07.02 | robin_z | probably trying to forward to 911 |
| 21:07.04 | Seldon1975 | wouldn't a call forwarded to '91' just not connect? |
| 21:07.08 | robin_z | Hmmhesays: yeah right |
| 21:07.13 | Hmmhesays | especially if you're just looking to forward vmail's |
| 21:07.34 | robin_z | Hmmhesays: Id just like exim to send mail out to people ... |
| 21:07.39 | Seldon1975 | robin_z: regardless of the intention, how could a call be connected to '91' |
| 21:07.45 | [TK]D-Fender | mishehu : look for "persist" in sip.cfg and you'll see all 3 |
| 21:07.50 | Hmmhesays | you can't get exim to send out mail? |
| 21:07.56 | robin_z | nope |
| 21:07.58 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:07.59 | robin_z | sodding thing |
| 21:08.16 | Hmmhesays | what'd you do? |
| 21:08.19 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
| 21:08.34 | robin_z | echo "this is a test" | xmail -s "test mail" foo@bar.com |
| 21:08.36 | Seldon1975 | so, noone can see how the call could have been forwarded to '91'? |
| 21:08.51 | brad_mssw | Seldon1975: all depends on your dialplan and what that actually means |
| 21:09.29 | Seldon1975 | brad_msswq: 9 is the number used to get an outside line |
| 21:09.40 | robin_z | Hmmhesays: I know jack shit about exim, Ive always used qmail myself |
| 21:09.41 | Seldon1975 | brad_msswq: but 91 isnt explicitly mentioned |
| 21:10.58 | Seldon1975 | does call forwarding happen when users use the COnference feature of their phones? |
| 21:11.04 | *** join/#asterisk tgrman (n=tgrman@picard.ojc.nuvio.com) |
| 21:11.33 | *** part/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 21:11.43 | sevrdseot | Why wouldn't my FOB be responding? |
| 21:11.46 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 21:13.26 | brad_mssw | Seldon1975: not forwarding ... but you could have 2 FXOs and 1 SIP or FXS or IAX bridged in a 3 way |
| 21:14.21 | *** join/#asterisk techy (n=tecky@66.9.96.115) |
| 21:15.09 | Seldon1975 | brad_mssw: ok thanks |
| 21:16.02 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 21:16.18 | *** join/#asterisk pigpen2 (n=mark@65.91.220.162) |
| 21:16.19 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 21:16.54 | oceanlan | anyone have a good VoIP traffic simulator that i could download? |
| 21:17.14 | Seldon1975 | brad_mssw: still not sure how they could be connected to '91' |
| 21:18.24 | Alric | Hmmm. Good timing I guess. Was about to put a production PBX in with 1.2.2 :) |
| 21:18.41 | denon | use 1.2.1 :) |
| 21:18.55 | Alric | 1.2.1 crashes when I use voicemail. |
| 21:19.05 | denon | heh, then you've got other problems :) |
| 21:19.11 | Alric | So why not 1.2.3? |
| 21:19.13 | fugitivo | Alric: that's weird |
| 21:19.22 | Alric | Yep. Full seg fault. |
| 21:19.29 | fugitivo | Alric: too new for a production system, 1.2.2 was new too |
| 21:19.36 | denon | go ahead and use 1.2.3, just test everything out as you normally should .. its young |
| 21:20.03 | Alric | Yeah |
| 21:20.04 | fugitivo | i never moved from 1.2.1 |
| 21:20.18 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 21:20.24 | Alric | I wouldn't have, except every time I hit VoicemailMain I got a seg fault. 1.2.2 fixed it. |
| 21:20.36 | denon | permissions issue? |
| 21:20.42 | denon | though I cant imagine permissions doing that |
| 21:20.52 | Alric | Don't think so, nothing changed between the upgrade really. |
| 21:21.04 | denon | well, yeah - but 1.2.2 could handle the permissions problem better |
| 21:21.06 | buzzyd | Is there anyway to set meetme so the admin user can mute all other users or all non admin users? |
| 21:21.26 | Alric | I noticed asterisk CLI got a nice warranty disclaimer message. |
| 21:21.44 | *** join/#asterisk cidez (n=sdgseg@modemcable159.198-81-70.mc.videotron.ca) |
| 21:21.54 | furiousgeorge_ | how dows this iaxprov.conf file work w/o asking you for the passwords? |
| 21:22.51 | bkw__ | mac addr |
| 21:23.27 | furiousgeorge_ | bkw__ was that to me? |
| 21:24.49 | denon | yes |
| 21:26.07 | jhiver | mhhh |
| 21:26.08 | furiousgeorge_ | denon: is there anything wrong with just working it into my dialplan, say, goto(s-$DIALSTATUS) and use provider on congenstion or chanunavail |
| 21:26.24 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:26.25 | furiousgeorge_ | *use alt_provider that is |
| 21:26.28 | jhiver | question about voicemail |
| 21:27.10 | jhiver | how do you make asterisk say "busy" when you're using voicemail rather than give you this "bip bip" saying you have another call waiting? |
| 21:27.16 | sevrdseot | When I navigate my browser to AMP I get 172.16.8.3:9050, but when I click on FOP I get 172.16.8.3/panel and it says Connection Refused. How would I connect to the panel? |
| 21:27.32 | jhiver | am I even making sense? |
| 21:27.38 | jhiver | I check my voicemail |
| 21:27.45 | jhiver | somebody calls |
| 21:28.42 | jhiver | I want my extension to be considered "busy" |
| 21:28.42 | file | well, you can either write dialplan logic to allow only one call to your phone... |
| 21:28.42 | file | or, it's up to your phone to send the signal back it's busy |
| 21:28.42 | jhiver | mhhh |
| 21:28.45 | jhiver | I don't think that my phone supports call waiting |
| 21:28.48 | jhiver | so I think it's asterisk doing it |
| 21:28.53 | file | is it zaptel, or SIP? |
| 21:29.02 | *** join/#asterisk Iam8up|lappy (n=dontemai@cpe-71-65-112-38.woh.res.rr.com) |
| 21:29.07 | jhiver | SIP -> Fritz!FonBox -> DECT phone |
| 21:29.14 | znoG | guys, i have a Digium TDM400P card with 4 x FXO modules. I can't get rid of the echo problems. I've read' the info on voip-info.org about echo cancellation, tried ztmonitor and adjusting rxgain/txgain settings to no avail. |
| 21:29.24 | jhiver | rather, Asterisk -> SER -> Fritz!FonBox -> DECT phone |
| 21:29.27 | file | if it's SIP, all that logic and stuff is in the SIP device connected to Asterisk |
| 21:29.35 | file | Asterisk just sends another call to you |
| 21:29.47 | jhiver | well this is kind of strange |
| 21:29.54 | Iam8up|lappy | anyone know of a good (free?) network test that's designed to see how well voip will run? theres some good looking software named sip sim and 'gl packetgen'... |
| 21:29.56 | jhiver | because here is what happens |
| 21:30.05 | sevrdseot | file: would you happen to know why it tells me that the connection to my panel is refused? |
| 21:30.10 | jhiver | when I am on a "normal" call, I never ever have this behavior |
| 21:30.16 | jhiver | but when I am on voicemail, I do |
| 21:30.27 | file | sevrdseot: I do not support AMP, and neither do a lot of people in here - try #amportal |
| 21:30.35 | file | jhiver: pastebin the console output |
| 21:30.38 | sevrdseot | Sorry. |
| 21:30.48 | jhiver | file: ok hang on |
| 21:31.10 | brad_mssw | sevrdseot: read the docs ... most likely you're not running the op_server.pl daemon, which FOP requires to be running |
| 21:31.12 | jhiver | it'll take a couple of minutes |
| 21:31.28 | harryvv | znog, are those the cards with the new echo chips on it? |
| 21:31.43 | znoG | harryvv: have no idea, it was purchased a while ago. |
| 21:31.56 | harryvv | whats a while ago? |
| 21:32.11 | *** join/#asterisk signaleleven (n=evan@lion.ragga-jungle.com) |
| 21:32.14 | fugitivo | harryvv: no |
| 21:32.21 | znoG | harryvv: about.. 2 years or so |
| 21:32.22 | harryvv | Iam8up|lappy there are a few |
| 21:32.26 | fugitivo | harryvv: the tdm2400p comes with echocan, not the tdm400p |
| 21:32.41 | harryvv | znog, okay the new cards have some echo supression technoligy on them. |
| 21:32.43 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 21:32.52 | Iam8up|lappy | harryvv - can you name any of them for me, please? are they free/open source? |
| 21:33.13 | harryvv | Iam8up|lappy do a google search. or voip-info.org. I dont recall the addresses. |
| 21:33.16 | fugitivo | znoG: check the wiring |
| 21:33.21 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:33.27 | znoG | harryvv: i see, and if I don't have those chips? :) |
| 21:33.35 | fugitivo | znoG: i have several tdm400p and i never had to modify the rx/tx gain |
| 21:33.38 | znoG | fugitivo: telephone wiring? all that part is OK, have checked. |
| 21:33.40 | *** join/#asterisk dalabera (n=dala@pmr.pmrtechnologies.com) |
| 21:33.47 | znoG | fugitivo: you just leave rxgain and txgain at 0? |
| 21:33.50 | fugitivo | znoG: yes |
| 21:33.58 | harryvv | znog, this is normally a impedence issue |
| 21:34.05 | znoG | fugitivo: do you have an asterisk box handy with a tdm400p? |
| 21:34.16 | fugitivo | yes |
| 21:34.17 | znoG | fugitivo: i'm just wondering if the fact that its sharing an IRQ with the ethernet card could be the cause of it |
| 21:34.24 | fugitivo | znoG: sure |
| 21:34.26 | fugitivo | change the irq |
| 21:34.28 | Seldon1975 | for outgoing calls I have the following rule: exten=>_9.,1,Dial(Zap/g1/${EXTEN:1}) which uses available trunkfs from group 1 to make the call; can I instead specify a prioritized list of the Zap channels to use in order? |
| 21:34.30 | sevrdseot | brad_mssw: op_server _is_ running, asterisk 5051 3.3 2.5 7704 6472 pts/3 S 16:33 0:01 /usr/bin/perl -w ./op_server.pl |
| 21:34.33 | *** join/#asterisk _cleric_ (n=dacleric@p5482B24E.dip0.t-ipconnect.de) |
| 21:34.35 | znoG | fugitivo: i thought you might be able to check if /proc/interrupts lists the tdm400p sharing an IRQ |
| 21:34.40 | harryvv | fugitivo I dont think so |
| 21:34.49 | dalabera | hi, have 1.2.0 and something weird is being happening with the calls.. does it might be related the critical update? |
| 21:35.00 | jhiver | file, there http://pastebin.ca/38596 |
| 21:35.00 | fugitivo | znoG: you should never share the irq of your card |
| 21:35.08 | dalabera | I've searched on bugs.digium.com but haven't found anything... |
| 21:35.09 | harryvv | impedence mistmatch in the pstn lines can cause signal bounce back. |
| 21:35.31 | jhiver | I basically have a cron job that everyminutes : 1) check if I have new messages 2) check if I am available |
| 21:35.41 | fugitivo | harryvv: sure, but changing rx/tx gain will kill software echocan |
| 21:35.42 | Iam8up|lappy | harryvv - can you give me any NAMES? google comes up with software that i'd have to purchase... |
| 21:35.49 | jhiver | if 1) and 2) are satisfied, it generates a call file and voicemail calls me |
| 21:35.50 | file | jhiver: back to what I was saying, Asterisk doesn't do the call waiting for SIP - it just sends another call |
| 21:35.59 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 21:36.09 | jhiver | ok... |
| 21:36.12 | jhiver | mhhh |
| 21:36.17 | jhiver | ok :) |
| 21:36.32 | *** part/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 21:36.40 | Math[laptop] | there's no such thing as "call waiting" in voip, since all phones can handle many calls at once, call waiting is what's given to analog phones |
| 21:36.43 | harryvv | Iam8up|lappy I would buy it if it is what you need. |
| 21:37.02 | jhiver | I guess I'll have to find a way for the script to "know" if there is already an instance of the voicemail calling me to avoid generating another .call file |
| 21:37.12 | Iam8up|lappy | harryvv - i'd prefer not ot buy something when theres a free alternative =P |
| 21:37.19 | harryvv | Math, I get the call waiting beep on our polycom but, most calls come in on one line anyway. |
| 21:37.45 | Math[laptop] | harryvv: you surely get the beep, but thats the phone who generates it |
| 21:37.56 | file | jhiver: groups |
| 21:38.08 | jhiver | file, please explain! |
| 21:38.09 | jhiver | :) |
| 21:38.51 | file | use chan_local to loop your call through the dialplan instead of going direct, and use groups to only allow 1 call to your phone at a time |
| 21:39.16 | Math[laptop] | heh chan_local is so useful |
| 21:39.24 | file | under certain circumstances yeah it is |
| 21:39.43 | jhiver | I guess I have to look chan_local up then |
| 21:39.44 | jhiver | thanks |
| 21:39.44 | pifiu | in IAX, whats the difference between user= and username= ? |
| 21:39.46 | kink0 | I get: status is 'CHANUNAVAIL' for my Zap , E1 , any idea ? |
| 21:40.03 | file | http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup example 5 |
| 21:40.03 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:40.08 | *** join/#asterisk Trazz (n=traderz@ip-66-80-141-13.chi.megapath.net) |
| 21:40.17 | file | or 7 |
| 21:40.20 | kink0 | of course I get also: T4XXP (PCI) Card 0 Span 1 OK |
| 21:40.23 | jhiver | file: thanks! |
| 21:40.27 | kink0 | so appears may be ok. |
| 21:40.29 | file | you'll get the drift... |
| 21:40.48 | funxion | b4 I try this can anyone tell me if Dial(Zap/g6/) will just go off hook on a chnnel assigned to that group and conect the call without spilling digits? |
| 21:40.48 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de) |
| 21:41.00 | kink0 | at the trunk I got something like this: pxx/pxx--> id.ind !setup (0004) BEA-CC-Sp-C-64k-ALaw CHA-Int-P-1-01 |
| 21:41.00 | kink0 | <PROTECTED> |
| 21:41.02 | harryvv | Math[laptop] yea? how do I put the current person on hold and then answer the other call? this is a IP500 |
| 21:41.17 | pifiu | hold |
| 21:41.18 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 21:41.22 | pifiu | transfer or blind transfer |
| 21:41.23 | pifiu | extension |
| 21:41.29 | pifiu | oh |
| 21:41.36 | pifiu | no just hold, scroll down to the other call and pickup |
| 21:41.36 | *** join/#asterisk lookitsme (n=lookitsm@d198-166-20-237.abhsia.telus.net) |
| 21:41.43 | Alric | Do Setgroup and Checkgroup work? |
| 21:41.50 | Math[laptop] | hardwire: you press answer of the phone..... |
| 21:42.01 | Iam8up|lappy | harryvv - http://www.voip-info.org/wiki-How+To+Debug+and+Troubleshoot+VOIP <--- =) |
| 21:42.11 | Math[laptop] | s/hardwire/harryvv/ |
| 21:43.10 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 21:43.21 | pifiu | in IAX, whats the difference between user= and username= ? |
| 21:43.23 | jsaunders | Is it {$EXTEN:2} to strip first 2 chars? I can't remember. |
| 21:43.27 | harryvv | laptop, I never have looked into the cw feature...not that is should be for this type of phone. Once I get my DID working then will assign each extention on this phone for each phone number and then it should not be a issue. |
| 21:43.30 | [av]bani | yes, :2 |
| 21:43.34 | jsaunders | tnx |
| 21:43.38 | fugitivo | jbot: aaa eee iii |
| 21:43.47 | fugitivo | s/aaa/.+/ |
| 21:43.50 | Math[laptop] | jsaunders: tho its ${EXTEN:2} not {$EXTEN:2} |
| 21:43.58 | lookitsme | must go today 1 alienware area51-m 5700 laptop price 650 includes shipping, carry case. message me on mcsltd@telusmail.net on msn or on mikcomputing on aim |
| 21:44.05 | fugitivo | s/.+/./g |
| 21:44.06 | file | lookitsme: no. |
| 21:44.13 | fugitivo | s/.+/./ |
| 21:44.32 | jsaunders | Ah yes. :) |
| 21:44.35 | jsaunders | Heheh |
| 21:44.39 | *** join/#asterisk hans (n=fugalh@hfugal.NMSU.Edu) |
| 21:44.57 | fugitivo | lookitsme: can you send it to Argentina? |
| 21:45.19 | svenna_ | man, i finaly got it up and running again.... |
| 21:45.46 | Math[laptop] | lol |
| 21:46.13 | tzanger | heh I started my weights again yesterday and I'm starting to feel it today |
| 21:46.34 | tzanger | I'd like to think that my muscles are growing again but I know they're just a big welt right now :-) |
| 21:47.09 | lookitsme | must go today 1 alienware area51-m 5700 laptop price 650 includes shipping, carry case. message me on mcsltd@telusmail.net on msn or on mikcomputing on aim |
| 21:47.17 | Math[laptop] | shut up |
| 21:47.19 | fugitivo | lookitsme: can you send it to Argentina? |
| 21:47.28 | fugitivo | lookitsme: if not, stop being a spambot |
| 21:47.32 | fdask | heh |
| 21:47.35 | file | fugitivo: he does this a lot |
| 21:47.37 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 21:47.41 | fugitivo | file: ban him! |
| 21:47.45 | tzanger | it's probably hot |
| 21:47.47 | file | I can't |
| 21:47.50 | fugitivo | damn |
| 21:47.58 | Dr-Linux | http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords |
| 21:48.00 | Dr-Linux | exten => *122,7,GotoIf($[["${result}" = "0"]?20:30) |
| 21:48.02 | kink0 | hi Dr-Linux |
| 21:48.03 | file | but I can do this. |
| 21:48.04 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-227-78.claranet.co.uk) |
| 21:48.05 | Dr-Linux | whats wrong with this |
| 21:48.06 | fugitivo | let's pingdeath him |
| 21:48.10 | jlewis | thats one high end laptop...seems awfully cheap |
| 21:48.16 | Dr-Linux | kink0 :) how are you |
| 21:48.21 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 21:48.44 | kink0 | fucked ... unable to get running my PRI E1 |
| 21:48.44 | tzanger | Dr-Linux: you have two [[ |
| 21:48.52 | tzanger | kink0: still?? |
| 21:49.02 | kink0 | tzanger, now the problem is other :) |
| 21:49.06 | pifiu | iax2 is a pitaa..... |
| 21:49.23 | Dr-Linux | tzanger: WIKI is wrong :P |
| 21:49.30 | Katty | wikiwikiwikiwiki mushroom mushroom |
| 21:49.32 | kink0 | tzanger, the problem is I get a SIP 503 "Service Unavailable" |
| 21:49.32 | *** mode/#asterisk [+b *!n=lookitsm@*.abhsia.telus.net] by twisted[asteria] |
| 21:49.33 | *** kick/#asterisk [lookitsme!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted[asteria] (stop) |
| 21:49.38 | tzanger | Dr-Linux: so fix it |
| 21:49.39 | kink0 | or 600 Denied |
| 21:49.39 | file | that's what I can do :) |
| 21:49.39 | Math[laptop] | Katty: lol |
| 21:49.43 | fugitivo | yeaaaaahhhh |
| 21:49.52 | Katty | twisted[mobile]: aroo? |
| 21:50.04 | kink0 | and I see at the trunk side some messages that I am unable to googled |
| 21:50.15 | Katty | oh. |
| 21:50.24 | kink0 | like : pxx/pxx<++ id.req !relcompl (0004) CAU-elemerror |
| 21:50.26 | Assid | umm.. if i setvar in 1 context.. can i use that in another context? |
| 21:50.40 | sevrdseot | is there a place in asterisk for a SIP user to record a personalize outgoing voice mail message? |
| 21:50.52 | kink0 | tzanger, but at least, at the moment I got my zap show status lists as OK |
| 21:51.08 | tzanger | kink0: is the D channel up? |
| 21:52.01 | kink0 | yes !! now is up... *CLI> == Primary D-Channel on span 1 up |
| 21:52.17 | tzanger | good |
| 21:52.26 | tzanger | that means for the most part that the PRI signaling is good |
| 21:52.28 | Hmmhesays | old asterisk install where nat is not being detected properly |
| 21:52.33 | Hmmhesays | still have the private ip addy showing up |
| 21:52.48 | znoG | fugitivo: yea, i didn't purposely share it, but it seems to be shared by the eth card |
| 21:53.04 | znoG | fugitivo: 11: 29932853 XT-PIC wctdm, wctdm, eth0 |
| 21:53.17 | znoG | fugitivo: strange thing is after the first few seconds of the call, the echo is gone |
| 21:53.23 | kink0 | tzanger, yes, i think signalling is fine, I get OK for the interface, as well I can see some debug messages at the 2N Stargate PRI side |
| 21:53.36 | kink0 | but unable to understand why is not working. |
| 21:54.10 | tzanger | what exactly is the stargate |
| 21:54.19 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 21:56.38 | kink0 | tzanger, is a GSM gateway, that has a PRI E1 interface |
| 21:56.57 | fugitivo | znoG: try moving the card to another slot to not share the irq, and the echo dissapearing after 5 seconds is the echotraining working, what settings do you have in zapata.conf? |
| 21:57.06 | tzanger | kink0: ok, and where exactly does SIP come into play |
| 21:57.10 | dextro | does anyone here understand the bug this morning and can explain it to me;after looking at the patch i dont understand how the date of the machine factors in |
| 21:57.34 | tzanger | dextro: every 2^22 seconds a condition arises where a computation results in a number unable to be handled. |
| 21:57.50 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:57.57 | kink0 | tzanger, ok, I orginate the call from machine1 ( Asterisk ) to machine2 ( Asterisk ) . Machine2 has the PRI connected to the Stargate gateway PRI |
| 21:58.17 | tzanger | kink0: ok, and machine2 is giving the SIP error to Mache1? |
| 21:58.20 | tzanger | er Machine1 |
| 21:58.22 | kink0 | both, machine2 and Stargate reports ok and up her PRI interfaces, and I see the dialog |
| 21:58.26 | *** part/#asterisk furiousgeorge_ (n=email@pool-70-111-13-159.nwrk.east.verizon.net) |
| 21:58.54 | kink0 | tzanger, yes, but error is originated in some side between machine2 and Stargate communication |
| 21:59.02 | tzanger | no |
| 21:59.13 | tzanger | stargate is talking ISDN to *, there are no SIP error codes coming from the stargate. |
| 21:59.30 | dextro | tzanger: and that condition appears constantly after this morning at whatever time? is this stuff in channel.c ? |
| 21:59.31 | kink0 | tzanger, machine2 reports -- Called r1/657897123 |
| 21:59.31 | kink0 | <PROTECTED> |
| 21:59.33 | tzanger | kink0: when Machine2 gets a request, what does the CLI show? |
| 21:59.52 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 21:59.54 | tzanger | kink0: that tells me that the ISDN switchtype is likely not right |
| 21:59.58 | kink0 | tzanger, can I paste you in priv. ( are just about 5 lines ) |
| 22:00.10 | tzanger | pri debug span 1 should give you a TON of infomration that is not only confusing but also cryptic. :-) |
| 22:00.10 | kink0 | to avoid abuse on the public room |
| 22:00.15 | tzanger | kink0: just pastebin it |
| 22:00.17 | tzanger | ~pastebin |
| 22:00.18 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:00.38 | kink0 | ok... going to pastebin site |
| 22:01.05 | tzanger | kink0: I have to go now though |
| 22:01.14 | tzanger | my sweetheart takes priority to this stuff I'm afraid. :-) |
| 22:01.19 | pifiu | pastebin |
| 22:01.57 | znoG | fugitivo: i'll pastebin it |
| 22:02.00 | znoG | ~pastebin |
| 22:02.01 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:02.14 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 22:03.23 | kink0 | http://pastebin.com/523003 |
| 22:03.39 | kink0 | tzanger, ok... thanks. |
| 22:04.02 | znoG | fugitivo: http://pastebin.com/523004 |
| 22:04.26 | tzanger | X-Asterisk-HangupCause: Invalid information element contents |
| 22:04.28 | tzanger | kink0: ^^^ |
| 22:04.38 | tzanger | your GSM gateway is rejecting *'s ISDN signaling |
| 22:04.46 | drumkilla | kick it in the face |
| 22:04.49 | drumkilla | that usually works |
| 22:04.57 | tzanger | try Set(CALLERID(all)=) to clear it .. I've seen that mess it up |
| 22:05.11 | tzanger | you may also want to turn facilityenable=no in zapata.conf on machine2 and re*START* asterisk |
| 22:05.38 | tzanger | but yeah this could get nasty. make absolutely sure that your signaling is the same (euroisdn or qsig is my guess, but it oculd be damn near anything) |
| 22:05.51 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 22:06.01 | *** join/#asterisk johnm__ (n=johnm@80.189.248.169) |
| 22:06.14 | redder86 | so is NuFone down to the timebomb or is it some other reason? |
| 22:06.23 | kink0 | tzanger, may be this : Called r1/657897123 ??? |
| 22:06.25 | tzanger | redder86: it wouldn't be the timebomb... must be something else |
| 22:06.36 | tzanger | kink0: doubtful. it's saying it doesn't like the IEs |
| 22:06.39 | kink0 | appears gateway is claiming about received an invalid called number |
| 22:06.40 | dextro | tzanger, do you have a second to explain how that "2^22 seconds condition" effects calls only after this morning? |
| 22:06.52 | tzanger | kink0: pri debug span 1 and place *1* call and pastebin the crapload of data that comes up on machine2 |
| 22:06.58 | tzanger | kink0: pri no debug span 1 to turn it off |
| 22:07.00 | wunderkin | redder86: there was an email blast sent out (with poor english) that their ETR would be around 6PM EST |
| 22:07.01 | tzanger | but I will have to catch it later |
| 22:07.03 | tzanger | I'm late now :-) |
| 22:07.06 | kink0 | tzanger, yes, that was reporting while pri debug |
| 22:07.19 | tzanger | kink0: I need to see it on the machine with the PRI |
| 22:07.24 | tzanger | /var/log/asterisk/debug I think |
| 22:07.31 | redder86 | wunderkin: yeah, I got it. Was poorly written, and made me wonder if they got hit with the timebomb. But apparently it was just coincidence. |
| 22:07.47 | *** join/#asterisk erieasterisk (n=asterisk@67.129.205.6) |
| 22:07.51 | kink0 | may be possible that asterisk send as number r1/1234567 instead just 1234567 to the gateway ? |
| 22:07.55 | pifiu | what causes when one makes a call to not have a ringback tone? |
| 22:07.58 | wunderkin | no idea, i dont use nufone now anyway so i dunno |
| 22:08.13 | pifiu | like just dead air but have it be rining in the other side? |
| 22:08.38 | oceanlan | omfg ! my * box just crashed updating perl!! wow, 333mhz sux! |
| 22:08.48 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 22:08.51 | erieasterisk | hello folks |
| 22:08.57 | erieasterisk | damn, that does suck oceanlan |
| 22:09.14 | }btorch{ | what happens if I installed * on another location and then I installed amp ? |
| 22:09.20 | oceanlan | hahaha...i guess thats what I get for loding up an old ass IBM!! |
| 22:09.30 | rob0 | why wouldn't a 333mhz box crash on updating perl? |
| 22:09.33 | rob0 | I mean ... |
| 22:09.38 | rob0 | why WOULD |
| 22:09.54 | pifiu | dude IAX gets complicated |
| 22:09.56 | tainted_ | rob0 b/c it's an OLD ASS IBM DUH |
| 22:09.57 | oceanlan | its still running...i think...hard drive is cranking, but my SSH's locked up |
| 22:10.24 | oceanlan | <}btorch{>: i think you will dick your entire box if you try that! |
| 22:10.37 | rob0 | <== has run a lot of old machines which were/are stable |
| 22:10.37 | oceanlan | I think i tried it once... |
| 22:10.42 | }btorch{ | installing amp or asterisk on another location ? |
| 22:11.01 | oceanlan | yea, this one has been good...this is like the 14th install!! we have been playing with betas alot |
| 22:11.10 | }btorch{ | I've had asterisk installed on /opt/asterisk for quite sometime now its fine |
| 22:11.11 | De_Mon | anyone successfully able to play h726 voicemail video with eyebeam? |
| 22:11.26 | oceanlan | wow, maybe i did something different... |
| 22:11.41 | }btorch{ | amp is giving me crap though cause it looks for the default paths |
| 22:11.47 | oceanlan | all i know is that AMP takes control of the DB's and apache... |
| 22:11.59 | oceanlan | yea, i think thats the problem we had... |
| 22:12.00 | rob0 | My MX server is a PII-350, rock solid. |
| 22:12.12 | }btorch{ | do you like amp ? |
| 22:12.29 | oceanlan | ohh..and BTW, dont ever CTRL+C the amp install...it finger f*cks the mysql users/pass's |
| 22:13.02 | oceanlan | this 333mhz has worked solid for a while..i think perl h8s ibm |
| 22:13.12 | }btorch{ | i wasn't even able to install it until remebered to set the old-passowrd on mysql my.cnf |
| 22:13.43 | }btorch{ | now everything seems fine but the panel which doesn;t work |
| 22:13.49 | oceanlan | we wnet thru that once too |
| 22:14.06 | erieasterisk | i am running asterisk with 4 analog phones and a TDM400P (4 FXS modules) card. We have a minor issue. |
| 22:14.11 | rob0 | oceanlan: compiling perl? That would be a big job for a little CPU, for sure. I try not to compile much on the old boxes. |
| 22:14.20 | pifiu | what causes when one makes a call to not have a ringback tone? |
| 22:14.32 | oceanlan | Amp requires perl though... |
| 22:14.35 | znoG | fugitivo: see anything wrong with it? |
| 22:14.41 | dextro | can someone explain how the "2^22 seconds condition" effects calls only after this morning? or just in general how that bug "works" and what the patch is fixing. point me to a web site if one is available, i have read the release notes,changelogs and the mantis bug reports but still have a lot of space to fill |
| 22:15.02 | rob0 | Doesn't your distro provide perl? |
| 22:15.16 | oceanlan | negative...CentOS 4.2 server |
| 22:15.29 | rob0 | hmmmm, strange. |
| 22:15.45 | rob0 | oh well .. good luck with it |
| 22:15.51 | oceanlan | meh, its still running...it pings...hard drive is going nutz! but no SSH ability as of yet... |
| 22:15.58 | oceanlan | yea, if anything...FORMAT! |
| 22:16.08 | rob0 | give it time to finish, it might still recover. |
| 22:16.25 | *** part/#asterisk johnm (n=johnm@80.189.248.169) |
| 22:16.33 | Dr-Linux | exten => *122,7,GotoIf($["${result}" = "0"]?20:30) |
| 22:16.37 | oceanlan | hehehe...hope so! supposed to do a demo with this batch tomorrow!~ |
| 22:16.44 | Dr-Linux | something wrong in this line :S |
| 22:17.22 | erieasterisk | we have 4 phones (4 lines) in our office and lets say i am on a call and the other phones ring, i hang up and i try to answer the call from my phone and it will not pickup. It's because the ring command was already issued..I know..however is there a way for the asterisk to ring the phone right after i hang it up? |
| 22:17.34 | znoG | Dr-Linux: looks OK to me |
| 22:18.25 | znoG | erieasterisk: you could put both phones in the same pickup group and do a *8# |
| 22:19.13 | Dr-Linux | hhm.. |
| 22:19.14 | erieasterisk | what does the *8# exactly do? |
| 22:19.25 | [av]bani | incinerates your pc |
| 22:19.29 | Dr-Linux | znoG: then it's another problem, please have a look |
| 22:20.42 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 22:23.36 | *** part/#asterisk thosa (n=thosa@p54879AE2.dip0.t-ipconnect.de) |
| 22:25.11 | *** join/#asterisk zmb (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 22:25.16 | erieasterisk | hello znoG. thank you for your response |
| 22:25.26 | zmb | hello all |
| 22:25.30 | erieasterisk | however, what does *8# exactly do? |
| 22:25.42 | pifiu | can someone help me out with this? |
| 22:25.43 | pifiu | http://pastebin.ca/38608 |
| 22:25.52 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 22:25.56 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 22:26.15 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:26.16 | znoG | erieasterisk: it picks up a ringing call from the pickup group |
| 22:26.26 | *** join/#asterisk jarnaud (n=ja@c-67-191-4-38.hsd1.fl.comcast.net) |
| 22:26.27 | znoG | erieasterisk: search voip-info.org for call pickup |
| 22:26.38 | zmb | i need help setting up asterisk to register as sip client using realm and auth |
| 22:26.43 | erieasterisk | awesome!!!! |
| 22:26.51 | erieasterisk | thank you so much. it worked like a charm! |
| 22:26.59 | erieasterisk | my boss will be very happpy. |
| 22:27.02 | erieasterisk | :) |
| 22:27.10 | jarnaud | Hi all, I was at a Mark Spencer presentation and he mentionned a TOPSECRET directory where to find funny audio files. Anyone know where is it? |
| 22:27.12 | tehdely | is there anything available for asterisk that does very basic sound-matching |
| 22:27.18 | zmb | does anyone have any help file etc that shows the way to do it |
| 22:27.18 | tehdely | say, i wanted to automate stepping through someone else's ivr system |
| 22:27.28 | tehdely | and expect certain known prompts |
| 22:27.34 | file[laptop] | jarnaud: the CD of doom |
| 22:28.06 | jarnaud | ;) |
| 22:28.21 | pifiu | where is that directory? |
| 22:28.23 | Hmmhesays | i hate it when customers get upset when I don't get them a concrete answer if something is going ot work or not |
| 22:28.26 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 22:28.34 | Hmmhesays | especially when they can't make a freaking test call |
| 22:28.40 | jarnaud | It's a funny Allison message saying "... you are not the next personn in line..." |
| 22:29.06 | pifiu | oh that directoy |
| 22:29.09 | pifiu | its just in the sounds |
| 22:29.14 | pifiu | nothing topsecret |
| 22:29.33 | tehdely | anybody know? <- |
| 22:29.36 | jarnaud | no it's a 2 minutes soundfile Mark is using in his presentations |
| 22:29.49 | tehdely | "We're out gambling, and getting drunk" |
| 22:29.53 | tehdely | ^ favorite allison sample |
| 22:30.02 | zmb | hehe.. did anyone even see my messages? |
| 22:30.33 | znoG | erieasterisk: you just tested it and it worked? |
| 22:30.36 | pifiu | which directoy? |
| 22:30.40 | pifiu | or its not out yet? |
| 22:30.56 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:30.59 | Hmmhesays | zmb about what? |
| 22:31.22 | zmb | hmmhesay about sip.conf register/auth using realm |
| 22:31.38 | De_Mon | I've got an extension with a NoOp(${CALLERID}) line, but I don't see it printed on the CLI |
| 22:31.43 | erieasterisk | yes, i just tested it cause i am at work and sure enough, it worked! |
| 22:31.50 | erieasterisk | thanks |
| 22:31.55 | zmb | need help !!! cant find the right way to do it.. and cant find any sample or docs that talk about it |
| 22:31.57 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-72-247.mia.bellsouth.net) |
| 22:32.06 | Derkommissar | Hey peeps, good afternoon |
| 22:32.07 | jarnaud | pifiu: someone in the audience asked Mark where to find this sound and he answered that it's available somewhere in a topsecret subdir |
| 22:32.23 | Derkommissar | I been reading up on channel_bluetooth |
| 22:32.33 | Derkommissar | and i think its pretty cool |
| 22:32.39 | Derkommissar | but not well documented |
| 22:33.27 | Hmmhesays | zmd: what do you mean using realm? |
| 22:34.16 | Dr-Linux | is there anyway, that user can change their own passwords in authenticate application? or DISA application? |
| 22:34.28 | websae | Does anyone know of a good company that sells ATA adapters to VoIP service providers that also sell a the provisioning for it as well? |
| 22:34.34 | Derkommissar | so maybe someone can awnser this question, can a call be place from a sip phone -> Asterisk -> Chan_bluetooth -> cellphone -> Central office -> cell phone ????? |
| 22:34.50 | Hmmhesays | Dr-Linux:use a passcode file and call an agi to change it |
| 22:34.57 | Derkommissar | websaw my favorite ata's are the worldacc |
| 22:34.58 | pifiu | well when you guys find it let me know |
| 22:35.01 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:35.02 | websae | worldacc |
| 22:35.07 | pifiu | btw can someone help me with this? http://pastebin.ca/38608 |
| 22:35.10 | websae | do they have provisioning and monitoring equipment for that? |
| 22:35.15 | Derkommissar | yes |
| 22:35.21 | websae | do you have a link? |
| 22:35.24 | Hmmhesays | worldacc, lol |
| 22:35.25 | websae | i would greatly appreciate it |
| 22:35.26 | Hmmhesays | fark them |
| 22:35.45 | KranZ | websae: ABP technology |
| 22:35.53 | KranZ | 972-831-1600 |
| 22:35.57 | websae | ABP technology? |
| 22:35.59 | websae | is that you? |
| 22:36.02 | KranZ | no |
| 22:36.05 | Hmmhesays | all points bulletin? |
| 22:36.16 | KranZ | you mean all bulletin points? |
| 22:36.24 | Hmmhesays | Dr-Linux: slip me a 50 and i'll whip you up a script to do it |
| 22:36.26 | websae | is that the number for ABP? |
| 22:36.36 | KranZ | yeah |
| 22:36.37 | Hmmhesays | 911? |
| 22:36.37 | websae | Derk: what's the link for worldacc? |
| 22:36.45 | Derkommissar | one sec |
| 22:36.56 | KranZ | www.abptech.com |
| 22:37.06 | Derkommissar | http://www.worldaccxx.com/ |
| 22:37.15 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 22:37.15 | Hmmhesays | I have all the autoprov software for them |
| 22:37.18 | websae | thanks! |
| 22:37.19 | Hmmhesays | we used to be a dealer |
| 22:37.19 | KranZ | i know they'll provision the linksys devices |
| 22:37.20 | zmb | hmmhesays http://www.voip-info.org/wiki-SIP+Authentication describes it a bit.. but no examples |
| 22:37.21 | KranZ | np |
| 22:37.35 | Derkommissar | np |
| 22:37.45 | Derkommissar | i think that their ata is the best |
| 22:38.04 | Hmmhesays | zmb: what do you need to know? |
| 22:38.08 | Derkommissar | fax support works like a charm |
| 22:38.10 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 22:38.18 | KranZ | websae: i havent used them, but my sales contact at linksys referred them |
| 22:38.23 | Derkommissar | so anyone here played with asterisk and bt ? |
| 22:38.23 | Dr-Linux | Hmmhesays: passcode file is there, but i want user can change/set there passwords? :S |
| 22:38.27 | Hmmhesays | i'm fond of mediatrix, cause i've used them forever |
| 22:38.39 | Hmmhesays | Dr-Linux: like i said slip me a 50 and i'll write you a script to do it |
| 22:38.41 | Hmmhesays | ;D |
| 22:38.49 | Dr-Linux | :@ |
| 22:38.50 | zmb | hmmhesays how to correctly use it. i have a sip account with a provider and am able to use it using x-lite softphone. want to use asterisk |
| 22:39.10 | Hmmhesays | what settings do you set in xlite? |
| 22:39.12 | De_Mon | do I need a certain debug level to see NoOp's on the CLI? |
| 22:39.15 | *** join/#asterisk NeonLevel (n=NeonLeve@dsl-201-133-181-39.prod-infinitum.com.mx) |
| 22:39.21 | Hmmhesays | set verbose 3 |
| 22:39.28 | Dr-Linux | Hmmhesays: you mean its not possible with authenticate app? :) |
| 22:39.32 | Hmmhesays | if memory serves |
| 22:39.42 | Hmmhesays | you see a flag on the wiki for it Dr-Linux? |
| 22:39.48 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 22:40.01 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:40.18 | zmb | hmmhesays... plug in xyz.host.com for domain/realm... and node.xyz.host.com:9999 for sip proxy |
| 22:40.35 | zmb | hmmhesays plus the account info. |
| 22:40.37 | Dr-Linux | Hmmhesays: no |
| 22:40.39 | Hmmhesays | ahh zmb, yo uare confusing yourself |
| 22:40.45 | Hmmhesays | Dr-Linux: then prolly not |
| 22:40.54 | De_Mon | Hmmhesays 3 works, didn't realise debian set verbosity to 0! |
| 22:40.59 | Dr-Linux | hhm.. |
| 22:41.07 | *** part/#asterisk erieasterisk (n=asterisk@67.129.205.6) |
| 22:41.13 | Hmmhesays | verbosity is set with whatever flags you start asterisk with |
| 22:41.23 | Hmmhesays | zmb, use the register line |
| 22:41.42 | Hmmhesays | register => password:username@blahbittyblah.com |
| 22:41.45 | Dr-Linux | sip port 5060 can be changed? most of internet provider blocked this port in my country |
| 22:41.56 | Hmmhesays | or is it the username:password |
| 22:42.09 | Hmmhesays | you can initiate a sip call on any port you want |
| 22:42.12 | Hmmhesays | but you can only listen on one |
| 22:42.33 | elvisthedj | looking for troubleshooting advice: Came home today, and I have no audio with any of my 3 providers (one sip, two iax). I can dial in to * and hear the menu, send the dtmf, but when the call is bridget.. silence on both ends regardless of which provider i go through. I've made no router changes, so i'm not sure where to sart looking. I've tried dialing my fxs connected via sip/ulaw and my softphone (also sip) and both gi |
| 22:42.39 | jaike | dr-linux: that sucks..what country? |
| 22:42.41 | Dr-Linux | Hmmhesays: and the one is? |
| 22:42.49 | zmb | i tried. problem is that domain/realm is not the same as sip proxy host.. |
| 22:42.53 | Dr-Linux | jaike: Pakistan |
| 22:42.53 | Hmmhesays | Dr-Linux: whatever one you want that is not in use |
| 22:42.53 | _Sam-- | elivs: its your version |
| 22:42.59 | _Sam-- | there are bugs in 1.2.2or3 |
| 22:43.03 | *** join/#asterisk Defraz (n=t0tal@72.165.56.43) |
| 22:43.04 | _Sam-- | what are you running |
| 22:43.14 | elvisthedj | _Sam--: cvs head from about 4 days ago |
| 22:43.20 | _Sam-- | check topic |
| 22:43.20 | elvisthedj | _Sam--: it worked last night :( |
| 22:43.23 | Hmmhesays | i had to deal with some pakistani's today, not fun |
| 22:43.23 | elvisthedj | oh |
| 22:43.25 | zmb | hmmhesays looking at the log... sip debug and xlite... message headers are not hte same... |
| 22:43.25 | elvisthedj | :D |
| 22:43.25 | elvisthedj | k |
| 22:43.38 | Dr-Linux | hh... |
| 22:43.45 | websae | Kranz: do you know how much their equipment cost? |
| 22:43.45 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 22:43.53 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:43.54 | websae | Derk: do you know how much worldacc equipment costs? |
| 22:43.59 | Hmmhesays | zmb: what is the difference? |
| 22:44.00 | Dr-Linux | so is it good approach to change port 5060 ? |
| 22:44.04 | Hmmhesays | the realm? |
| 22:44.15 | KranZ | websae: nope, u check the website? |
| 22:44.36 | websae | yeah |
| 22:44.37 | websae | i did |
| 22:44.42 | zmb | hmmhesays as message should go to host for sip proxy but user info is for domain .. sip.provider.com is domain/realm... node.sip.provider.com is sip proxy... and they are different ip |
| 22:44.45 | websae | they don't offer provisioning equipment |
| 22:45.00 | Hmmhesays | you don't register to a proxy |
| 22:45.33 | *** join/#asterisk Psykick (n=anon@smtp.phoenixone.co.nz) |
| 22:45.35 | Dr-Linux | what else i'd need to change if i change SIP port 5060 in sip.conf ? |
| 22:45.43 | Hmmhesays | to listen? nothing |
| 22:45.50 | KranZ | websae: i havent used them because we havent needed their services, but my sales contact at linksys mentioned their provisioning |
| 22:45.59 | Psykick | mike are you here? |
| 22:46.00 | bsdfreak | heh |
| 22:46.14 | bsdfreak | can anyone recommend a voip-capable phone that also supports analog lines for a mixed telephony setup? |
| 22:46.17 | websae | hrm |
| 22:46.19 | websae | what do you use? |
| 22:46.26 | bsdfreak | me? |
| 22:46.29 | bsdfreak | asterisk |
| 22:46.37 | bsdfreak | and i have two pots lines |
| 22:47.02 | bsdfreak | i'm using a sipura 3000 now which is decent, but the fxo support is awful |
| 22:47.15 | Hmmhesays | then you have it setup wrong |
| 22:47.19 | bsdfreak | heh |
| 22:47.23 | jaike | it youre to receive a call from a voip provider, shouldnt they reconfigure their end as well? to send to another port, say 5061? |
| 22:47.36 | bsdfreak | hmmhesays: how do i eliminate dial tones during conversations then? |
| 22:47.45 | bsdfreak | they seem to be triggered by voice, and it seems to be a fairily known problem |
| 22:47.50 | *** join/#asterisk hroarke (n=whois@216.190.144.90) |
| 22:47.52 | Hmmhesays | dialtones? |
| 22:47.55 | bsdfreak | i've tried lowering the incoming and outgoing voice volume |
| 22:47.56 | bsdfreak | yeah... |
| 22:47.58 | bsdfreak | sorry |
| 22:47.59 | bsdfreak | dtmf |
| 22:48.05 | bsdfreak | long day. heh |
| 22:48.14 | Hmmhesays | explain the issue |
| 22:48.27 | bsdfreak | it's randomly interpreting conversation during calls received on the fxo port as dtmf codes |
| 22:48.37 | bsdfreak | so someone says something in a certain pitch and volume and it echos a dtmf |
| 22:48.42 | Dr-Linux | jaike: what do you mean i didn't understand explain please? |
| 22:48.55 | bsdfreak | trying to pass the dtmf from the fxo port to asterisk |
| 22:49.03 | Hmmhesays | bsdfreak: the fxo side hears the dtmf or the ip side |
| 22:49.21 | bsdfreak | it appears to hear it from either one |
| 22:49.27 | KranZ | wow, whoever can speak dtmf is talented |
| 22:49.27 | cyburdine | hey guys (and gals?) is there a way to enable an outbound proxy setting to connect to a VOIP provider? |
| 22:49.37 | Hmmhesays | are you using inband dtmf perhaps? |
| 22:49.37 | cyburdine | in sip.conf? |
| 22:49.40 | KranZ | they'd make for a great phreaker |
| 22:49.42 | bsdfreak | as in it'll hear it from someone on the pots side and relay it to the voip and vice versa |
| 22:49.44 | hroarke | hi there :) does anyone know the link to the digium developer kit on the digium site, I would like to purchase it but cant find the link anymore? |
| 22:49.47 | _Sam-- | lol |
| 22:49.47 | *** join/#asterisk oceanlan|dustin (n=info@cpe-71-65-112-38.woh.res.rr.com) |
| 22:49.50 | bsdfreak | hmmhesays: i've tried every possible config w/ and w/o inband |
| 22:50.10 | Hmmhesays | i've never had that problem, do you have a lot of convo's with female dwarfs? |
| 22:50.12 | _Sam-- | would be good for handicap people...voice activated dtmf |
| 22:50.14 | bsdfreak | no |
| 22:50.16 | KranZ | "i have a natrual ability to phreak. i speak dtmf." |
| 22:50.19 | bsdfreak | this happens with females and males alike |
| 22:50.27 | KranZ | natural even |
| 22:50.31 | Hmmhesays | how often does it happen? |
| 22:50.33 | bsdfreak | ok there captain crunch |
| 22:50.41 | bsdfreak | hmmhesays: several times per call |
| 22:50.45 | cyburdine | i keep getting a registration retry... and am assuming that is it since all my other info is correct |
| 22:50.45 | bsdfreak | sometimes several times per minute |
| 22:50.50 | Hmmhesays | and you can duplicate this at will? |
| 22:51.04 | KranZ | bsdfreak: you sure its not crosstalk? |
| 22:51.07 | bsdfreak | not at will, but it could probably be recorded |
| 22:51.08 | _Sam-- | considering its a known problem, first time ive heard anyone with it here (i am by no means an expert) |
| 22:51.16 | bsdfreak | kranz: yeah, absolutely sure. |
| 22:51.22 | bsdfreak | doesn't happen on voip to voip or pots to pots |
| 22:51.24 | Hmmhesays | plug your fxs into your fxo port and call yourself |
| 22:51.37 | bsdfreak | heh |
| 22:51.48 | bsdfreak | sam: i've seen it on forums before |
| 22:51.54 | _Sam-- | i believe ya |
| 22:51.55 | *** join/#asterisk foosbar (n=ffd@65.205.244.100) |
| 22:52.04 | bsdfreak | hmm: heh |
| 22:52.07 | *** join/#asterisk Kohat (n=Nothing@202.125.141.6) |
| 22:52.07 | _Sam-- | what card are you u sing? |
| 22:52.08 | KranZ | isnt there a relaxed dtmf setting? |
| 22:52.14 | bsdfreak | i'm not, it's a sipura 3000 |
| 22:52.27 | bsdfreak | kranz: that doesn't help if you arne't using an fxo card on the asterisk server |
| 22:52.29 | Hmmhesays | the sipura shouldn't be generating dtmf tones |
| 22:52.50 | KranZ | Hmmhesays: it should be passing them as info |
| 22:53.00 | Hmmhesays | or rfc2833 |
| 22:53.05 | Dr-Linux | sorry dc |
| 22:53.13 | Dr-Linux | jaike: what do you mean i didn't understand explain please? |
| 22:53.22 | bsdfreak | heh |
| 22:53.23 | KranZ | bsdfreak: have you done a sip debug and seen if the sipura is sending dtmf info |
| 22:53.25 | bsdfreak | i believe it is |
| 22:53.34 | bsdfreak | yeah, and iirc it is |
| 22:53.56 | Hmmhesays | iirc and believe it is are not solid statements |
| 22:54.04 | KranZ | does not compute |
| 22:54.16 | Dr-Linux | Hmmhesays: what else i will need to change if i changed the 5060 port? |
| 22:54.18 | Hmmhesays | either it is or it isn't |
| 22:54.29 | Hmmhesays | the listening port on all your endpoints |
| 22:54.34 | KranZ | bsdfreak: during the call the sipura is sending dtmf info when you hear the tones? |
| 22:54.41 | cyburdine | anyone know how to define a different outbound proxy than the incoming sip proxy? |
| 22:54.44 | bsdfreak | i can't remember i'll have to enter in to a debug |
| 22:54.59 | KranZ | yeah, only debug the ip of the sipura |
| 22:55.00 | bsdfreak | let me try to do it now |
| 22:55.03 | bsdfreak | yea |
| 22:55.05 | *** join/#asterisk seelen (n=_seele@200.124.172.72) |
| 22:55.25 | bsdfreak | heh |
| 22:55.32 | *** join/#asterisk |cleric| (n=dacleric@p5482824A.dip0.t-ipconnect.de) |
| 22:55.38 | KranZ | bsdfreak: you might need to set dtmf to info on the sipura |
| 22:55.44 | seelen | Hi there i need some help with this: In my company there are already several complains about their calls being hanged up with no reason |
| 22:55.45 | Hmmhesays | or not use info |
| 22:55.58 | bsdfreak | i've tried all the different dtmf settingss |
| 22:56.00 | bsdfreak | no diff |
| 22:56.07 | KranZ | yeah, just for this test |
| 22:56.18 | KranZ | to see if the sipura is the culprit |
| 22:56.26 | foosbar | Speaking of the sipura 3000, am I better off using it's FXO port to talk to an SBC line, or getting a digium card? |
| 22:56.26 | pifiu | ok my dear friends |
| 22:56.31 | pifiu | i need help on this plz |
| 22:56.35 | pifiu | http://pastebin.ca/38608 |
| 22:56.45 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:56.57 | Hmmhesays | ugh i need a cigarette like now |
| 22:57.10 | bsdfreak | hmm |
| 22:57.11 | KranZ | sux to be addicted |
| 22:57.16 | bsdfreak | it appears to be creating a native bridge |
| 22:57.24 | Hmmhesays | between what? |
| 22:57.28 | KranZ | zap to sip? |
| 22:57.33 | bsdfreak | the sipura and my extension |
| 22:57.35 | Hmmhesays | Kranz:sucks to work in IT |
| 22:57.43 | bsdfreak | Attempting native bridge of SIP/206-4d31 and SIP/200-b0f0 |
| 22:57.45 | KranZ | Hmmhesays: indeed |
| 22:57.50 | wunderkin | seelen: what version of asterisk? |
| 22:57.51 | KranZ | oh, sip to sip |
| 22:58.01 | KranZ | i thought the problem was sip to pstn |
| 22:58.08 | bsdfreak | it is |
| 22:58.13 | bsdfreak | but the pstn is on the sipura |
| 22:58.19 | KranZ | ohh |
| 22:58.21 | bsdfreak | and it converts it to sip when it transmits it to asterisk |
| 22:58.22 | bsdfreak | heh |
| 22:58.28 | KranZ | ic now |
| 22:58.39 | bsdfreak | debug isn't showing anything |
| 22:58.47 | bsdfreak | when i hit a dtmf on the phone itself |
| 22:58.51 | bsdfreak | it says attempting native bridge |
| 22:59.01 | Hmmhesays | no iax account in the terminating iax box |
| 22:59.21 | bsdfreak | but when the tone happens "from a voice" it doesn't say anything |
| 22:59.21 | wunderkin | yaya bsdfreak dont worry about it |
| 22:59.25 | bsdfreak | yeah? |
| 22:59.28 | seelen | wunderkin, 1.20 |
| 22:59.31 | bsdfreak | i should worry about it, it's interrupting calls |
| 22:59.34 | bsdfreak | and making a nuisance |
| 22:59.36 | bsdfreak | heh |
| 22:59.42 | Hmmhesays | and I think your iax syntax is farked |
| 23:00.02 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 23:00.02 | Hmmhesays | IAX2/username@<host>/extension |
| 23:00.11 | bsdfreak | heh |
| 23:00.32 | wunderkin | oh, ok.. that.. |
| 23:00.37 | bsdfreak | it would appear then |
| 23:01.06 | wunderkin | bsdfreak: i had that problem a couple times.. someone suggested that i change to rfcblah dtmftype.. |
| 23:01.11 | seelen | wunderkin, 1.2.1.. is there an issue or bug in particular in this version that causes that hanging problem?? |
| 23:01.17 | bsdfreak | wunderkin: and did it work? |
| 23:01.41 | wunderkin | bsdfreak: well i only had that problem a couple times, on the same day.. and it hasnt happened since so i dunno |
| 23:01.48 | bsdfreak | only choices are AVT and INFO |
| 23:01.54 | wunderkin | seelen, so it went from 1.2.0 to 1.2.1 in 2 minutes? |
| 23:02.00 | fugitivo | seelen: 1.2.1 is ok |
| 23:02.03 | KranZ | you got latest firmware on both sip devices? |
| 23:02.08 | bsdfreak | and one can add "inband" to each choice |
| 23:02.10 | wunderkin | ya i think that bug only affects 1.2.2 |
| 23:02.13 | NeonLevel | hello, i have an asterisk server and two sip clients using linksys pap2-na (both behind diferent linux masquerading) the diagram is in http://kyo.gdlicanet.net.mx/~otto/images/linksys.jpg my question is how can i get the media (rtp) travel directly from pap to pap, without using the asterisk. (i have already tried the canreinvite and does not seem to work) can aynone please give me a clue????? please check the diagram.... |
| 23:02.25 | wunderkin | bsdfreak, it may be avt, ive heard it before but not sure what it is |
| 23:02.34 | bsdfreak | wunder: are you using the sipura 3000? |
| 23:02.38 | wunderkin | no |
| 23:02.43 | KranZ | bsdfreak: latest firmware? |
| 23:02.49 | De_Mon | gah, I go to the trouble of registering for the wiki only to find out it was already correct :D |
| 23:02.52 | [TK]D-Fender | bsdfreak : You want to use INFO and set it as such in sip.conf |
| 23:02.59 | fugitivo | NeonLevel: you can't if devices are on different networks |
| 23:03.11 | seelen | wunderkin, sorry i was confused about it so i checked it out |
| 23:03.28 | *** join/#asterisk [ToTo] (n=ToTo@host42-157.pool874.interbusiness.it) |
| 23:03.53 | seelen | wunderkin, but then the real problem could be something disabled.. that hangs up calls??.. they hang up in no particular timing |
| 23:04.09 | NeonLevel | fugitivo: thanks for your help, is there any device that might do something like this? |
| 23:04.43 | bsdfreak | kranz: yup |
| 23:04.43 | *** join/#asterisk hans (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:04.46 | bsdfreak | tk: i'll give it a shot |
| 23:04.47 | wunderkin | seelen, maybe you have a different problem then |
| 23:05.26 | seelen | wunderkin, different from what??'.. i dont understand what you say? |
| 23:05.31 | [TK]D-Fender | bsdfreak : I run an SPA-3000 here. |
| 23:05.48 | bsdfreak | cool appreciate the advice |
| 23:06.24 | wunderkin | seelen, than what i was thinking.. if your problem is reproducable maybe you should try upgrading to the latest 1.2 release branch |
| 23:06.44 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 23:06.47 | bsdfreak | heh |
| 23:06.51 | bsdfreak | just set it up accordingly |
| 23:06.53 | bsdfreak | gonna test again now |
| 23:06.59 | *** join/#asterisk backblue_ (n=moo@87-196-46-161.net.novis.pt) |
| 23:07.47 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 23:10.00 | fugitivo | i hate debian |
| 23:10.14 | fugitivo | that apt-get crap |
| 23:10.30 | Alric | yum! |
| 23:10.35 | fugitivo | i hate yum too |
| 23:10.52 | NeonLevel | what distro do you use fugitivo? |
| 23:10.57 | Alric | emerge! |
| 23:11.01 | fugitivo | gentoo or lfs |
| 23:12.06 | _Sam-- | man and just when we were getting along so good, you ahve to go start talking smack about debian! |
| 23:12.28 | fugitivo | lol |
| 23:12.33 | *** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 23:12.47 | _Sam-- | where did that comment come from, out of the blue..."i hate debian" |
| 23:12.51 | fugitivo | ok, i prefer debian before redhat, fedora and all that shit |
| 23:12.55 | _Sam-- | something much have triggered it |
| 23:13.51 | fugitivo | trying to fix something on a customer server and had to use "again" apt-get after years of not using it |
| 23:14.37 | _Sam-- | what package? |
| 23:14.51 | fugitivo | it doesn't matter, i already fixed it |
| 23:15.01 | De_Mon | Jan 25 18:14:18 WARNING[13307]: pbx.c:1690 pbx_extension_helper: No application 'verbose <AAAARGH: ${CALLERIDNUM}> 0' for extension (local, 300, 1) |
| 23:15.11 | De_Mon | verbose doesn't exist in 1.2.2? |
| 23:15.38 | fugitivo | De_Mon: there's a serious bug with 1.2.2, upgrade to 1.2.3 or downgrade to 1.2.1 |
| 23:16.32 | file | De_Mon: and your line in extensions.conf is wrong for that... |
| 23:16.55 | *** join/#asterisk bch (n=bch@CPE-70-92-133-175.mn.res.rr.com) |
| 23:17.11 | harryvv | so do these new digium cards with the echo cancelation boards make a big difference? |
| 23:17.24 | De_Mon | Usage: VERBOSE <message> [level] |
| 23:17.35 | De_Mon | I tried it with and without the <>'s and upper/lowercase verbose |
| 23:17.54 | file | you want this |
| 23:18.01 | file | verbose(<AAAARGH: ${CALLERIDNUM}> 0) |
| 23:18.09 | De_Mon | figures |
| 23:18.28 | De_Mon | **makes his first voip-info.org wiki edit! |
| 23:18.34 | Seldon1975 | hi |
| 23:18.41 | Seldon1975 | can someone please help me with my syntax |
| 23:19.00 | Seldon1975 | Im trying to get * to use certain trunk lines for ougoing calls, in priority order |
| 23:19.02 | Seldon1975 | so I have: |
| 23:19.32 | _Sam-- | fugitivo: have there ever been any performance comparisons of different linuxes? ie: is there any proof that one flavor is better than another at certain tasks? |
| 23:19.37 | Seldon1975 | exten=>_9.1,Dial(Zap/22&Zap/21&Zap/20/${EXTEN}) |
| 23:19.44 | Seldon1975 | but that doesnt work at all |
| 23:19.52 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 23:19.59 | harryvv | you dont need to use 9 in asterisk nessesarly |
| 23:20.04 | Seldon1975 | no |
| 23:20.09 | msw | _Sam--: benchmarks are only useful in determining how fast benchmarks run |
| 23:20.10 | Seldon1975 | thats irrelevent |
| 23:20.12 | De_Mon | _Sam-- different linuxes? do you mean distros? |
| 23:20.25 | _Sam-- | yeah, sorry, different stock distros |
| 23:20.39 | _Sam-- | like, is gentoo stock w/ apache faster than, say debian stock with apache package |
| 23:20.41 | De_Mon | heh, well the one you like the most runs just as fast as all the rest |
| 23:20.52 | De_Mon | _Sam-- are you factoring in compile time? |
| 23:21.06 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 23:21.12 | De_Mon | installing in debian is much faster than installing in gentoo |
| 23:21.16 | _Sam-- | im not factoring anything...im just wondering if such a comparison of any kind exists |
| 23:21.42 | _Sam-- | like is a stock debian installation with apache faster at loading web pages than a gentoo stock install w/ apache package, or any apples to apples comparisons |
| 23:22.03 | tehdely | debian has an ancient build of asterisk in apt |
| 23:22.07 | tehdely | i would suggest building your own |
| 23:22.19 | _Sam-- | they have 1.2.1 if you have the right sources lists |
| 23:22.29 | wunderkin | Seldon1975: umm... are you wanting it to dial one call out an available line? what that does is call out 3 lines .... you want a call group for that |
| 23:22.50 | Seldon1975 | aha |
| 23:23.03 | Seldon1975 | ok can I specify more than one group? |
| 23:23.13 | tehdely | anybody know of basic sound recognition package for asterisk |
| 23:23.16 | Seldon1975 | ie: it tries group 1 first, then group 2 |
| 23:23.17 | tehdely | i want to automate stepping through someone else's ivr |
| 23:23.21 | tehdely | and expect particular prompts, and respond to them |
| 23:23.27 | tehdely | i'm not looking for speeech recognition as much as simple sound matching |
| 23:23.29 | tehdely | anything like that? |
| 23:23.29 | Seldon1975 | wunderkin: ie: it tries group 1 first, then group 2 |
| 23:23.49 | _Sam-- | not that ancient: |
| 23:23.49 | _Sam-- | Package: asterisk |
| 23:23.49 | _Sam-- | Versions: |
| 23:23.49 | _Sam-- | 1:1.2.1.dfsg-3(/var/lib/apt/lists/ftp.de.debian.org_debian_dists_unstable_main_binary-i386_Packages)(/var/lib/dpkg/status) |
| 23:24.13 | *** join/#asterisk hd420 (n=hdiwan@c-69-181-3-188.hsd1.ca.comcast.net) |
| 23:24.31 | [TK]D-Fender | Slackware > ALL!!!! muahahaha *cough* |
| 23:24.43 | Alric | Oh no, not this again. |
| 23:24.47 | [TK]D-Fender | Slackware = Just works with * |
| 23:24.49 | [TK]D-Fender | ;P |
| 23:24.52 | _Sam-- | "i remember slackware 1.0" |
| 23:24.55 | Seldon1975 | Debian |
| 23:24.59 | _Sam-- | i have the cd around here someplace still |
| 23:25.00 | wunderkin | Seldon1975: well, yeah.. but all this depends on how you are doing everything..so |
| 23:25.10 | [TK]D-Fender | _Sam-- : You remember the Alamo too? ;) |
| 23:25.18 | Seldon1975 | wunderkin: know of any doco on the subject |
| 23:25.31 | Seldon1975 | wunderkin: of prioritising outgoing trunks |
| 23:25.35 | _Sam-- | getting that cd from walnut creek back then was a big deal. |
| 23:26.36 | Seldon1975 | my set up is pretty basic; I have 3 groups each with 2 Zap channels for my trunk lines |
| 23:27.00 | Seldon1975 | i just want to tell it to use group 1 before group 2 |
| 23:27.02 | Seldon1975 | etc |
| 23:28.30 | wunderkin | Seldon1975, what version of asterisk? |
| 23:28.54 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:29.08 | *** join/#asterisk Feral_Kid (n=kcc@red-corp-200.56.96.178.telnor.net) |
| 23:29.33 | Seldon1975 | wunderkin: 1.2 |
| 23:29.44 | wunderkin | i use groups on all the calls |
| 23:30.05 | wunderkin | if you could use groups of call groups that way would be easier but i dont think you can |
| 23:30.10 | [av]bani | i remember installing slackware from piles of floppies |
| 23:30.52 | Feral_Kid | [av]bani: Ah, those were the days... I can remember that it was a total of 33 3.5 floppies... |
| 23:30.55 | wunderkin | Seldon1975: GotoIf($[ ${GROUP_COUNT(g1)} > 22 ]?g2) Set(GROUP()=g1) etc |
| 23:30.56 | robin_z | piles of what? |
| 23:31.25 | Seldon1975 | wunderkin: any chance can I see your dialplan? |
| 23:31.38 | wunderkin | well that is all that would apply to it |
| 23:31.38 | brockj49464 | How do you get incoming calls from the same server/port to the correct extensions that your registered them with in v1.2.1 |
| 23:31.40 | robin_z | [av]bani: 8" or the modern 5 1/4" ones? |
| 23:31.57 | Seldon1975 | is it odd that prioritising your trunks should be so hard? strikes me as a fairly common need? |
| 23:32.13 | robin_z | Seldon1975: its not hard. its trivial |
| 23:32.15 | Feral_Kid | [av]bani: Assuming that one actually wanted to have TeX as well.. |
| 23:32.34 | Seldon1975 | robin_z: can you tell me how? |
| 23:32.41 | robin_z | Seldon1975: yes |
| 23:32.48 | wunderkin | Seldon1975: if you dont have the channels in other groups (in zaptel or zapata whatever it is) and you always want to do it that way on those channels, you can use call groups |
| 23:33.03 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 23:33.05 | [av]bani | [TK]D-Fender: spa-942 is sooooooooo disappointing. its even 10mb still! |
| 23:33.29 | wunderkin | Seldon1975: by call groups i mean using group= in zaptel or zapata whatever it is |
| 23:33.35 | [av]bani | Feral_Kid: downloaded them all over 14.4k dialup :) |
| 23:33.44 | *** join/#asterisk Netrunz (n=FF@195.137.136.242) |
| 23:33.50 | Netrunz | Hello everybody |
| 23:34.08 | Seldon1975 | wunderkin: yeah i have groups like that |
| 23:34.29 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 23:34.30 | Seldon1975 | robin_z: can you tell me how you approach this? |
| 23:34.35 | Feral_Kid | Ok, I have a slight problem... I am connected to my * box with a Sipura 2000. I think I am having a codec problem... When I try to dial VM or to an outside line, I hear absolutely nothing... Does this sound like a familiar problem? |
| 23:34.53 | Seldon1975 | robin_z: do you mean the way wunderkin is describing |
| 23:34.58 | robin_z | Seldon1975: what wunderkin said ... or, you can do it by the IsAvail or by creating a context for each trunk and then including them in order of preference in a master context or ... |
| 23:35.30 | robin_z | and then theres fallthrough ... |
| 23:35.45 | Feral_Kid | [av]bani: I was lucky... I was at college at the time, so I was on a T-1... But it was still a lengthy process!!! |
| 23:36.10 | Seldon1975 | wunderkin: thanks |
| 23:36.15 | Seldon1975 | robin_z: thanks |
| 23:36.28 | Netrunz | I have the following problem: My * machine takes calls from some sip clients and places them on a pstn line (with a wtdm). I want to make sure that the callerid visible to the outside is the one I decide and not the one declared by the clients (usually connected through a linksys pap2 ata) . How can I do that? :) |
| 23:36.43 | De_Mon | file is that verbose(message,level) or verbose(message|level)? |
| 23:36.51 | robin_z | Seldon1975: the trouble, if you like, with * is there are oftne severlal ways to solve a problem .. the tools are three, its just working out which ones to use and when ... |
| 23:37.00 | robin_z | three? |
| 23:37.03 | robin_z | there! |
| 23:37.05 | robin_z | sigh ... |
| 23:37.08 | De_Mon | robin_z that's a problem? |
| 23:37.09 | Seldon1975 | robin_z: heh |
| 23:37.10 | robin_z | cant tope. |
| 23:37.22 | robin_z | De_Mon: yes, yes it is. |
| 23:37.22 | Feral_Kid | Anyone have any ideas? |
| 23:37.23 | De_Mon | robin_z the trouble with math is there are often several ways to solve a problem... |
| 23:37.26 | Seldon1975 | robin_z: I looked up 'isavail' at voipinfo but i dont get any results |
| 23:37.37 | wunderkin | Seldon1975, chanisavail |
| 23:37.42 | Seldon1975 | aha |
| 23:37.44 | Seldon1975 | thx |
| 23:37.57 | wunderkin | show application chanisavail |
| 23:38.07 | wunderkin | easier than the wiki |
| 23:38.10 | Netrunz | I have the following problem: My * machine takes calls from some sip clients and places them on a pstn line (with a wtdm). I want to make sure that the callerid visible to the outside is the one I decide and not the one declared by the clients (usually connected through a linksys pap2 ata) . How can I do that? :) |
| 23:38.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:38.30 | robin_z | De_Mon: the tools are available and well documented. when to use them is less well documewnted and made worse by developer ego. |
| 23:38.31 | brockj49464 | Any ideas on how to get multiple accounts with same external PSTN server to work? |
| 23:39.14 | Netrunz | brockj49464 explain better |
| 23:39.17 | kink0 | is right that I must download from junghanns.net to fix EuroISDN PRI ? |
| 23:39.27 | Netrunz | kink0 yes |
| 23:39.30 | Err | Netrunz: you might consider waiting more than 2m before repeating your question |
| 23:39.40 | Netrunz | Ok Err sorry ;) |
| 23:39.50 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 23:40.13 | robin_z | De_Mon: by "developer ego" I mean the common thing where a developer makes some really funky, but obscure way of using some tool to do something quite normal in a weird way ... and then documents it inthe documentation, to massage their ego. Of course, they forget to document how to do it in the normal way as well .... |
| 23:40.31 | fiber0pti | Has anyone used sipX/ |
| 23:40.33 | brockj49464 | Netrunz: I have multiple accounts with the same provider but when they come in they are trying to connect to the wrong peer so they fail auth. |
| 23:40.39 | kink0 | Netrunz, I have the Cause: Unknown (100), class = Protocol Error (6) |
| 23:41.03 | kink0 | Netrunz, will junghanns.net fix this ? |
| 23:41.23 | Grubs | What is the status of +101 jump when busy for Dial commands? The wiki says for 1.2 its depreciated and you need to add the "j" option, yet old dial plans using +101 seem to work correctly. |
| 23:41.27 | Netrunz | kink0 what hardware r u using |
| 23:41.43 | harryvv | any html misteros here? |
| 23:41.45 | Netrunz | brockj49464 multiple accounts of?? |
| 23:42.21 | Netrunz | kink0 junghanns' patch is for isdn |
| 23:42.46 | Seldon1975 | robin_z, wunderkin: do you think the first example in http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail is going to give me what I need? |
| 23:43.14 | kink0 | Netrunz, I am ussing a Digium TE405 connected to a PRI E1 2N Stargate gateway |
| 23:43.17 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 23:43.44 | *** join/#asterisk choongii (n=choong@unaffiliated/choongii) |
| 23:43.57 | Netrunz | kink0 you need bristuff from junghanns |
| 23:44.07 | choongii | hello all :) am I in the right place for questions about zaphfc and bristuff? |
| 23:44.19 | Netrunz | choongii ask |
| 23:44.22 | Netrunz | :) |
| 23:44.34 | wunderkin | Seldon1975: yeah i guess so, the problem with these solutions is that the channel that is currently available could be used by something else before you dial out |
| 23:44.39 | choongii | great :) I'm very new to this, so please excuse my nonsense if I spout any |
| 23:44.59 | wunderkin | Seldon1975: which is why i use groups instead |
| 23:45.45 | Seldon1975 | wunderkin: I see - a race condition |
| 23:45.46 | brockj49464 | netrunz: http://pastebin.com/523142 is what I have that works but does not identify the correct incoming account but it is in the sip message. I can post that if you would like to see it. |
| 23:45.55 | choongii | I have a system with an ISDN card with a HFC chipset. if I understand correctly, after modprobing zaptel, zaphfc and qozap, I should be able to modprobe wcfxs. however, this command tells me that there is no such device |
| 23:46.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:46.58 | choongii | from what I understand, you need to have some kind of FXO and/or FXS module to make things work |
| 23:47.14 | choongii | +loaded |
| 23:47.25 | puzzled | evening all |
| 23:47.25 | wunderkin | Seldon1975: if you want i will message you what i use |
| 23:47.30 | choongii | evening puzzled |
| 23:47.48 | Netrunz | choongii you do not need all that. do you have bristuff installed? |
| 23:48.03 | litage | Sun designs their SPARC processors, but do they also make/fab them? |
| 23:48.08 | choongii | yes I have netrunz |
| 23:48.11 | Seldon1975 | wunderkin: that would be great |
| 23:48.13 | KranZ | anyone had an issue where you make a call from one sipphone using g726 to another exact sipphone which will only accept inbound at ulaw |
| 23:48.16 | Seldon1975 | wunderkin: thx |
| 23:48.17 | [av]bani | nope, sun sold off sparc mfg |
| 23:48.19 | KranZ | specifically spa-941 |
| 23:48.39 | KranZ | im trying to do g726 all the way |
| 23:48.41 | kink0 | Netrunz, I have downloaded bristuff, but appears are not for Digium cards... must I install it ? |
| 23:48.44 | [av]bani | sun has been following SGI into the path of self destruction |
| 23:49.06 | Netrunz | choongii the module yu need to modprobe is zaphfc.ko |
| 23:49.15 | brockj49464 | netrunz: Here it is with a sip debug http://pastebin.com/523149 |
| 23:49.19 | puzzled | anyone know if this the correct command to check out zaptel-1.2.2: svn checkout http://svn.digium.com/svn/zaptel/tags/1.2.2 zaptel-1.2.2 |
| 23:49.19 | Netrunz | choongii yes because it is for hfc |
| 23:49.32 | choongii | Netrunz, okay, thanks. I have it loaded -> zaphfc 9396 0 (unused) |
| 23:49.45 | KranZ | puzzled: svn co http://svn.digium.com/svn/zaptel/tags/1.2.2 zaptel-1.2.2 |
| 23:49.50 | kink0 | Netrunz, note I have a PRI ( not BRI ) |
| 23:49.56 | puzzled | KranZ: thanks |
| 23:50.19 | KranZ | puzzled: get asterisk 1.2.3, not 1.2.2 |
| 23:50.24 | Netrunz | kink0 yea i do too :) |
| 23:50.42 | KranZ | puzzled: zaptel and libpri are both at 1.2.2 still |
| 23:50.55 | Netrunz | choongii do ztcfg -vv |
| 23:51.03 | Netrunz | choongii tell me what it says |
| 23:51.12 | wunderkin | puzzled, it would be better to use the 1.2 release branch, details on asterisk.org download |
| 23:51.13 | Dr-Linux | sir few of my users can't register their X-Lite client to my server due to their internet provider blocked 5060 sip port, so whats the appropiate solution? any idea to change sip port for specific user? |
| 23:51.40 | puzzled | KranZ: yeah, read the announcement |
| 23:51.44 | choongii | Netrunz, the output I get is http://pastebin.com/523153 |
| 23:51.47 | KranZ | puzzled: good man |
| 23:52.04 | Netrunz | choongii |
| 23:52.09 | Grubs | Can anyone point me to documentation on the new 1.2 handling of return codes e.g. for Dial commands that replaces the old n+101 behaviour? |
| 23:52.10 | Netrunz | choongii should be ok now |
| 23:52.18 | Netrunz | choongii gotta do zapata.conf |
| 23:52.19 | puzzled | wunderkin: not if I diff 1.2 branch against 1.2.2 tarball cause I get a ton of .svn stuff in the diff |
| 23:52.24 | KranZ | anyone messed with the t38 hack? |
| 23:52.30 | kink0 | cvs http://www.asterisk-support.de/mirror/asterisk-1.2.2/asterisk-1.2.2.tar.gz ... hmmm must I edit install.sh and set to 1.2.3 ? |
| 23:52.35 | Netrunz | kranz i did |
| 23:52.37 | choongii | hmm alright.. no go though :( hold on |
| 23:52.40 | [TK]D-Fender | Grubs : Dial returns ${DIALSTATUS} |
| 23:52.44 | KranZ | Netrunz: get it compiled? |
| 23:52.54 | Netrunz | yea... |
| 23:52.55 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 23:53.03 | Netrunz | KranZ yea... |
| 23:53.10 | KranZ | Netrunz: from http://bugs.digium.com/view.php?id=5090 |
| 23:53.11 | KranZ | ? |
| 23:53.24 | Netrunz | KranZ yup! |
| 23:53.40 | Netrunz | KranZ bridging works pretty well |
| 23:53.48 | KranZ | which version did you patch |
| 23:53.52 | kink0 | Netrunz, install.sh appears to download again 1.2.2 and not 1.2.3, must I touch the install.sh script ? or just modprobe the new module ? |
| 23:53.59 | Netrunz | 1.2.1 |
| 23:54.01 | robin_z | sigh .. this poxy Zyxel .. how I hate thee oh crappy wifi phone! |
| 23:54.29 | Seldon1975 | Asterisk is comlaining that the command 'Cut' has been deprecated by 'CUT()' but if you do a search at voipware you get doco for Cut() |
| 23:54.35 | Netrunz | kink0 what install.sh script?? |
| 23:54.54 | kink0 | Netrunz, from bristuff |
| 23:55.10 | choongii | Netrunz, my zaptel.conf is here: http://pastebin.com/523154 and zapata.conf here: http://pastebin.com/523156 and the zap show channels output is here: http://pastebin.com/523159 .. could you please take a look for me, and let me know if things seem alright like this? because I am still getting the number-not-in-use-tone when I dial to it with my cellphone :( |
| 23:55.10 | Netrunz | oh allright |
| 23:55.10 | robin_z | Seldon1975: you'll never believe this, but there is more than one version of * in use .. |
| 23:55.10 | robin_z | Seldon1975: no, really. |
| 23:55.26 | kink0 | I did run install.sh as INSTALL says, but I observe download asterisk v 1.1.2 |
| 23:55.27 | Netrunz | kink0 i dunno if they work with the latest |
| 23:55.29 | KranZ | Netrunz: and my final question, have you made a t38 fax? |
| 23:55.50 | Seldon1975 | robin_z: but if CUT is the more current why is it not the DOCO default? |
| 23:55.59 | Netrunz | KranZ no only bridged one coming from a Grandstream 286 to a pstn fax |
| 23:56.00 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 23:56.01 | Seldon1975 | robin_z: and/or where is the doco for CUT |
| 23:56.11 | Grubs | thxs [TK]D-Fender!.. Goto(s-${DIALSTATUS},1) ... I see examples in extensions.conf now. Cheers :) |
| 23:56.22 | KranZ | yeah, so you were able to fax over ip to pstn using t38 |
| 23:56.33 | Netrunz | KranZ yes |
| 23:56.39 | KranZ | excellent |
| 23:56.41 | KranZ | thx |
| 23:56.49 | Netrunz | no prob ;) |
| 23:57.16 | Netrunz | <PROTECTED> |
| 23:57.56 | robin_z | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Cut |
| 23:58.00 | robin_z | ^^ there |
| 23:58.05 | robin_z | oh, its gone |
| 23:58.06 | KranZ | something like Set(CALLERID(number)=1234567) |
| 23:58.11 | KranZ | then dial |
| 23:58.44 | KranZ | 6pm, time to go home |
| 23:58.46 | Netrunz | KranZ i'll look it up on voip-info. r u sure it overwrites the clients' declared? |
| 23:58.51 | KranZ | yeah |
| 23:58.52 | brockj49464 | netrunz: I have done that via amp and the ext. |
| 23:58.57 | *** join/#asterisk hans (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:58.58 | KranZ | adios |
| 23:59.35 | [av]bani | hmm i want to hire don lafontaine to record my IVR menus |
