| 00:00.00 | Katty | ok kids! |
| 00:00.01 | iDunno | twisted[asteria] is lovely. I'm just upset because I got hit over the head with a fryingpan. |
| 00:00.04 | Katty | take this outside! |
| 00:00.04 | justinu | but I could understand why others might feel differently |
| 00:00.04 | iDunno | dammit. |
| 00:00.10 | twisted[asteria] | lol |
| 00:00.15 | twisted[asteria] | iDunno, :P |
| 00:00.33 | iDunno | (god I need sleep :) |
| 00:00.36 | twisted[asteria] | i joke i joke, i kid i kid |
| 00:01.25 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
| 00:01.49 | justinu | today has been a rough day |
| 00:01.51 | justinu | :( |
| 00:02.22 | *** join/#asterisk Oden (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 00:02.26 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 00:02.36 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 00:04.43 | twisted[asteria] | jumpin jesus on a pogo stick |
| 00:06.15 | riddlebox | hello, can someone help me with stream file in an agi script? |
| 00:06.28 | justinu | aww, shucks |
| 00:08.37 | riddlebox | I see in my console, that the script played the message, but I did not hear anything? |
| 00:09.42 | jpablo | unicall sucks, cisco sucks :'( |
| 00:12.57 | *** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.183) |
| 00:13.27 | justinu | i thought unicall was pretty cool |
| 00:15.39 | NeonLevel | good day everyone, listen i'm having the following scenario, i have two sips registers and here's what i want to do, i pick up one phone (zapata interface), and dial to one sip channel and works ok, when this call is being used, and i pick up other zapata channel to dial is there a way to try do dial this sip channel and if it fails then dial trhough the other sip channel? |
| 00:18.28 | NeonLevel | sorry... for my example, but i have no other way to put it out, besides my english is not as good as i wanted to be |
| 00:18.45 | *** part/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 00:18.51 | justinu | sounds like you want some kind of queue |
| 00:19.12 | NeonLevel | i'm not sure but perhaps that could be my solution |
| 00:19.21 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:19.31 | justinu | when you say "dial this sip channel and if it failes, then fial thru to the other sip channel" |
| 00:19.35 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 00:19.36 | NeonLevel | is there a way to put two sip channels in a queue? and then if one is busy, use the other? |
| 00:19.37 | Trazz | Ariel, you there? |
| 00:19.38 | justinu | you mean that if the phone is busy? |
| 00:19.54 | justinu | neon: you can also do this in the dialplan, with multiple Dial() commands |
| 00:19.54 | riddlebox | when I run my agi script, I get this, but I do not hear anything? and yes the menu is recorded? |
| 00:19.57 | justinu | and checking the DIALSTATUS |
| 00:20.33 | NeonLevel | ok, the problem is that i just want to dial the other sip channel only if the first one is busy |
| 00:20.44 | justinu | that's doable |
| 00:20.58 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 00:21.25 | Trazz | justinu, i cant get music on hold to work.. can you help |
| 00:21.31 | NeonLevel | only using the Dial() command or using the ChanIsAvail() command ? |
| 00:21.53 | NeonLevel | and thanks for your help justinu |
| 00:22.02 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 00:23.45 | justinu | http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 00:23.48 | justinu | for NeonLevel |
| 00:24.01 | Trazz | justinu, i cant get music on hold to work.. can you help |
| 00:25.05 | *** join/#asterisk HiltonT (n=HiltonT@office.quarkit.com.au) |
| 00:28.10 | justinu | Trazz: hang on |
| 00:28.38 | Trazz | ok |
| 00:31.41 | Trazz | whats ports do i need open my firewall for a softphone to connect to my asterisk server? |
| 00:31.51 | ManxPower | Mailing lists still down, I see. |
| 00:32.08 | riddlebox | Trazz:I believe that is in rtp.conf |
| 00:32.09 | ManxPower | Trazz, 5060 and whatever the phone uses for RTP. |
| 00:34.24 | Trazz | rtp is port 10000 through 20000 or is it port 10000 and port 20000 |
| 00:34.38 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 00:34.43 | riddlebox | 10000, and 20000 |
| 00:38.48 | Trazz | justinu still there? |
| 00:39.06 | justinu | yeah, busy here |
| 00:39.12 | *** join/#asterisk _nvrs (n=RUR@Kitchener-HSE-ppp3564307.sympatico.ca) |
| 00:40.29 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 00:42.25 | Trazz | so is the 5060 and rtpstart and end all UDP ports ? |
| 00:42.50 | riddlebox | can someone tell me why I wouldnt hear a file, when I am using a Stream file which has a file that exists |
| 00:44.31 | ManxPower | Trazz, you can set rtp.conf to be a smaller range of ports. |
| 00:44.35 | ManxPower | assume 2 ports per call |
| 00:49.23 | infinity1 | ok. now that my 601 plays nice --finally. what can i do with this 'browser'? |
| 00:51.09 | Mother | greetings |
| 00:51.18 | Mother | anyone here that uses chan_bluetooth? |
| 00:51.19 | *** join/#asterisk __nvrs (i=RUR@toronto-HSE-ppp4146183.sympatico.ca) |
| 00:55.35 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
| 00:55.40 | dily | hi@all |
| 00:56.15 | *** join/#asterisk Correa (n=Miranda@c906adc9.virtua.com.br) |
| 00:58.27 | syle | hi any iax proxy software? |
| 00:58.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 00:58.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:00.43 | *** join/#asterisk nvrs (i=RUR@HSE-Montreal-ppp3469548.sympatico.ca) |
| 01:01.35 | *** join/#asterisk Correa2 (n=Miranda@c906adc9.virtua.com.br) |
| 01:01.54 | *** join/#asterisk klictel (n=klictel@modemcable119.206-200-24.mc.videotron.ca) |
| 01:02.01 | *** join/#asterisk netcrash (n=andre@82.102.38.99) |
| 01:02.02 | Correa2 | Hi, I found a message on the mail list history about an opensource IP phone, possibly using ethernut. Does anyone knows about a project like that? |
| 01:02.23 | netcrash | Verb: '518', Identifier: '23', Endpoint: 'Unsupported', Version: 'or unknown' <- any ideas on what this means ? |
| 01:04.44 | netcrash | nevermind |
| 01:04.48 | netcrash | found the rfc |
| 01:05.06 | riddlebox | anyone want to help me with my stream file problem? |
| 01:05.07 | dily | stop now |
| 01:05.13 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 01:05.15 | dily | 8)))) |
| 01:05.44 | *** join/#asterisk annonimous (n=annonimo@201.137.44.223) |
| 01:05.49 | annonimous | hello |
| 01:05.51 | Trazz | my x-lite is not showing any voicemail messages yet i have 3 |
| 01:05.54 | riddlebox | dily, are you talking to me? |
| 01:06.00 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 01:06.01 | dily | no |
| 01:06.13 | CpuID | hmm...ok ppls got some little issues with hangup detection here...it seems fine 99% of the time, but every now and then a call comes inbound via a zap channel (digium tdm400p) that it just doesnt hangup |
| 01:06.14 | riddlebox | dily, whew |
| 01:06.27 | CpuID | now...just to add to it, theres an IVR system that loops on digit timeout |
| 01:06.27 | annonimous | just a little question qhere can i find a how to for enabling sipphone.com (aka gizmo) in my asterisk (im a noober but i want to learn to use asterisk) |
| 01:06.32 | CpuID | so it just sits there forever :) |
| 01:06.48 | CpuID | now...i decided to implement a little workaround, to force hangup after 3 loops of the main menu with no input |
| 01:07.09 | CpuID | which should solve the last 1% of issues...since i cant reproduce the problem locally to fix in zap :) |
| 01:07.21 | CpuID | sec let me pastebin |
| 01:07.33 | CpuID | something minor isnt quite right in my syntax |
| 01:07.40 | riddlebox | CpuID, do you know anything about AGI scripting? |
| 01:07.47 | CpuID | riddlebox, a little |
| 01:07.48 | CpuID | yea |
| 01:07.52 | CpuID | used it once or twice, whatya need? |
| 01:08.16 | riddlebox | CpuID, I am trying to use the stream file in the console it shows that everything went ok, but I dont hear anything |
| 01:08.29 | CpuID | stream file? |
| 01:09.59 | riddlebox | yeah you can play an audio file on a channel with it |
| 01:10.35 | ReD-MaN | hmm.. anyone have experience with Mitel SIP phones? just wondering how to get rid of having to press the dial button after punching in a number |
| 01:10.41 | riddlebox | http://pastebin.com/509133 is the little script and the output |
| 01:11.26 | riddlebox | hrmm I wonder if you cant play a .gsm file through it? |
| 01:11.33 | CpuID | ah...sec |
| 01:11.37 | CpuID | let me check it |
| 01:11.53 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 01:12.20 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 01:12.26 | riddlebox | CpuID:that line, showed be like this sys.stdout.write('STREAM FILE ToriMenu "*#"') |
| 01:12.31 | CpuID | sec |
| 01:12.46 | CpuID | http://rafb.net/paste/results/SGRdXu26.html |
| 01:12.58 | CpuID | if anyone has a solution, would be appreciated :) issue explained with example code |
| 01:13.06 | CpuID | ah STREAM FILE |
| 01:13.11 | CpuID | never used that myself |
| 01:13.43 | CpuID | is there a reason for using :gsm ? |
| 01:13.54 | riddlebox | I didnt mean for that to be in there |
| 01:13.58 | CpuID | hehe |
| 01:13.58 | CpuID | sec |
| 01:14.06 | riddlebox | sys.stdout.write('STREAM FILE ToriMenu "*#"') |
| 01:14.36 | CpuID | http://www.voip-info.org/wiki/view/stream+file |
| 01:14.36 | riddlebox | I have tried it with the full path it fails then, and also with the .gsm at the end |
| 01:14.37 | CpuID | STREAM FILE is known to behave inconsistently, especially when used in conjuction with other languages, i.e. Set(LANGUAGE()=xy). |
| 01:14.37 | CpuID | Workaround: Use EXEC PLAYBACK instead. |
| 01:15.15 | riddlebox | ahh I have been searching for two nights and never came up with that lol |
| 01:15.16 | CpuID | http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+AGI+MacinTalk&diff2=2 |
| 01:15.18 | CpuID | example |
| 01:15.18 | CpuID | :) |
| 01:15.55 | CpuID | ok...so anyone? http://rafb.net/paste/results/SGRdXu26.html |
| 01:17.37 | riddlebox | CpuID,replaced the stream file line with exec playback works great lol |
| 01:17.42 | CpuID | nice :) |
| 01:18.47 | riddlebox | now, I have a little more coding and my system will be done! |
| 01:19.06 | CpuID | heh, care to take a look at my issue btw? |
| 01:19.16 | CpuID | looking for some opinions atm...incase i missed something |
| 01:19.20 | riddlebox | I am but I think you are more advanced than me |
| 01:19.28 | CpuID | lol...its actually relatively simple :) |
| 01:19.37 | CpuID | i just havent worked with vars much in * configs myself |
| 01:21.27 | Trazz | justinu, back? |
| 01:21.28 | riddlebox | yeah I am learning all of this, it is basically my first project, and in it I am learning python,asterisk, and mysql |
| 01:22.03 | [av]bani | hmm.. gumstix looks ideal for SOHO pbx |
| 01:24.21 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 01:29.17 | CpuID | woo solved my own issue lol |
| 01:29.20 | CpuID | that code works :) |
| 01:29.28 | CpuID | was just not waiting for enough iterations :) |
| 01:30.05 | IMG-SD | QUESTION: |
| 01:30.05 | IMG-SD | Is is normal to receive an "Avoiding initial deadlock" message EVERY time either a Zap or Sip call is made??? See debug below: |
| 01:30.05 | IMG-SD | <PROTECTED> |
| 01:30.05 | IMG-SD | Jan 16 21:25:25 DEBUG[29224] chan_sip.c: Checking SIP call limits for device 1087 |
| 01:30.05 | IMG-SD | Jan 16 21:25:25 DEBUG[29224] chan_sip.c: build_route: Contact hop: <sip:1087@10.10.132.36:2054;line=1pp9al72> |
| 01:30.08 | IMG-SD | Jan 16 21:25:25 DEBUG[29219] channel.c: Avoiding initial deadlock for 'SIP/1087-7b9e' |
| 01:30.10 | IMG-SD | Jan 16 21:25:25 VERBOSE[30917] logger.c: -- Executing Dial("SIP/1087-7b9e", "SIP/1087|10") in new stack |
| 01:30.25 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-201.143.35.133.telnor.net) |
| 01:30.53 | IMG-SD | I get an Avoiding initial deadlock message about 2000 times a day!!!! Asterisk actually deadlocks once or twice a day on average. |
| 01:31.06 | dily | i have a problem when hangup, i receive this error Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx. any idea? |
| 01:31.09 | Trazz | how do you setup ivr? is there some good examples somewhere? |
| 01:31.42 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 01:34.18 | dily | . |
| 01:34.34 | *** part/#asterisk Correa (n=Miranda@c906adc9.virtua.com.br) |
| 01:34.51 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 01:36.37 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 01:38.04 | netcrash | can asterisk be used has a softswitch ? |
| 01:38.38 | fugitivo | asterisk is a pbx |
| 01:41.32 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 01:42.42 | [av]bani | some people do use it as a softswitch |
| 01:43.16 | tzanger | some people use a screwdriver as a hammer too... doesn't make it right |
| 01:43.43 | [av]bani | asterisk makes a better softswitch than a screwdriver makes a hammer |
| 01:43.51 | fugitivo | hell |
| 01:43.58 | fugitivo | why using asterisk as a softswitch? |
| 01:44.01 | fugitivo | it's a pbx |
| 01:44.07 | [av]bani | but elitism is fun |
| 01:44.20 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:44.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:44.25 | [av]bani | i guess if you want to be pedantic, buy a nortel |
| 01:44.28 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 01:44.44 | riddlebox | at least get a partner |
| 01:45.25 | fugitivo | maybe a nortel pbx will be a good softswitch |
| 01:45.39 | justinu | DMS500 |
| 01:45.56 | wunderkin | you have a few of those in your garage too justinu? |
| 01:46.09 | justinu | sort of |
| 01:46.29 | justinu | i could give you a tour of a DMS500 CO |
| 01:46.32 | justinu | what's left of it |
| 01:48.17 | justinu | we still have the main frames, the ama 9 track tape unit, and the gr303 stuff |
| 01:48.17 | justinu | but we sold off all the DTC/DTCIs |
| 01:49.04 | Trazz | justinu , do you have time to help with music on hold and ivr by chance |
| 01:49.19 | justinu | i don't do IVRs, but MoH maybe |
| 01:49.28 | Trazz | ok cool.. moh would be great :) |
| 01:49.33 | justinu | what's the problem? |
| 01:49.51 | Trazz | i call the extension and it goes to moh but no music |
| 01:49.58 | justinu | sip? |
| 01:50.02 | Trazz | ya |
| 01:50.08 | justinu | ah. known problem |
| 01:50.19 | justinu | did you answer the call? |
| 01:50.27 | tzanger | how will a norstar pbx be a good softswitch? |
| 01:50.42 | justinu | norstar is a ksu |
| 01:50.45 | Trazz | xten => 5000,1,Answer |
| 01:50.45 | Trazz | exten => 5000,2,MusicOnHold |
| 01:50.45 | justinu | meridian is a pbx ;) |
| 01:50.50 | Trazz | thats my exten |
| 01:50.52 | Trazz | the xtern is exten |
| 01:50.57 | justinu | trazz: hm. |
| 01:51.07 | Trazz | i call extension 5000 |
| 01:51.11 | Trazz | to test it |
| 01:51.13 | Trazz | it answers |
| 01:51.16 | Trazz | and dead air |
| 01:51.20 | justinu | trazz: if you type "rtp debug" on the console, then call, what do you get? |
| 01:51.26 | justinu | (paste it on pastebin) |
| 01:52.12 | justinu | you should see RTP packets going to and from your phone |
| 01:52.18 | Trazz | i get tons of these |
| 01:52.19 | Trazz | Got RTP packet from 192.168.1.94:8000 (type 3, seq 47, ts 40480, len 33) |
| 01:52.19 | Trazz | Sent RTP packet to 192.168.1.94:8000 (type 3, seq 31494, ts 26016, len 33) |
| 01:52.30 | Trazz | i hear a split second of music then silence |
| 01:52.33 | justinu | ok, so that is working fine |
| 01:52.44 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 01:52.50 | justinu | what kind of phone? |
| 01:52.58 | Trazz | x-lite |
| 01:53.22 | wunderkin | i would have bet money on that |
| 01:53.34 | justinu | weird i remember having a similar issue with eyebeam |
| 01:53.47 | wunderkin | silence suppresion |
| 01:53.55 | justinu | yeah, VAD or silence supression |
| 01:53.56 | Trazz | heheh any idea on how to fix it? |
| 01:54.01 | justinu | but i dunno if xlite supports that |
| 01:54.19 | wunderkin | transmit silence yes |
| 01:54.19 | justinu | trazz: verify those things are turned off in xlite |
| 01:54.26 | justinu | ahh |
| 01:54.33 | justinu | trazz: listen to wunderkin |
| 01:54.42 | *** part/#asterisk jbroome (n=jbroome@63-168-10-93.celito.net) |
| 01:54.45 | justinu | he's turning into a grizzled hardcore telecom expert |
| 01:54.53 | Trazz | eheheh let me see if i can find that |
| 01:55.23 | wunderkin | *pat pat justinu* |
| 01:55.49 | justinu | soon you'll have a DMS in your garage too |
| 01:56.20 | wunderkin | well, at the garage in my other house i do have some oc3, oc12, gigabit fiber switches (fore), and spools of multimode(?) fiber |
| 01:56.27 | justinu | sweet |
| 01:56.37 | wunderkin | oh, most of those are here |
| 01:56.47 | wunderkin | i havent been able to play with them yet |
| 01:56.47 | *** join/#asterisk IRS (n=Nbm@87-196-8-26.net.novis.pt) |
| 01:56.50 | justinu | you planning on lighting your block? |
| 01:56.55 | wunderkin | hehe |
| 01:57.00 | justinu | toss the fibre ober the back fence |
| 01:57.10 | justinu | hope they don't run over it with the lawnmower |
| 01:57.13 | wunderkin | i dont have anything to terminate them with yet :D |
| 01:57.28 | wunderkin | i just got it because it was cheap and local |
| 01:57.32 | justinu | cool |
| 01:57.42 | wunderkin | nerd factor |
| 01:58.15 | Trazz | awesome... it worked.. the volume is very low on it.. how can i turn it up ? |
| 01:58.31 | justinu | that's a problem with your audio device, not the music |
| 01:58.38 | Trazz | wunderkin, you rock ;) |
| 01:58.42 | justinu | softphones suck |
| 01:58.44 | wunderkin | i first got some of that stuff at my first apt out here, 2 of them are 150 lbs so one of them came in a freight truck, first time a freight truck has been to the apt complex, they laughed |
| 01:59.08 | justinu | i use to have a VME Sun workstation |
| 01:59.13 | justinu | it made a nice nightstand also |
| 01:59.24 | wunderkin | Trazz, thats either the volume of the sound files but more likely your pc if you are using the default files |
| 01:59.27 | wunderkin | hehe |
| 01:59.30 | justinu | also a space heater |
| 01:59.41 | Trazz | yes default files |
| 01:59.51 | justinu | trazz: they'll sound fine on a normal sip phone |
| 01:59.54 | wunderkin | i have some sparcstations, an ultra10? that isnt complete, and a partial alpha, an old hp |
| 02:00.04 | Trazz | ok |
| 02:00.19 | justinu | i used to have an sgi indigo2, but I gave that away about a year ago |
| 02:00.25 | wunderkin | heh |
| 02:00.45 | wunderkin | i had to get rid of a lot of crap too, been moving a lot |
| 02:00.56 | justinu | moving sucks too |
| 02:01.21 | justinu | i haven';t moved in a long time |
| 02:01.26 | justinu | not looking forward to doing that again |
| 02:01.32 | wunderkin | ;p |
| 02:03.27 | Trazz | wunderkin, do you know how to ivr with asterisk? |
| 02:03.44 | justinu | you can read about that on the net trazz |
| 02:03.47 | justinu | ~thebook |
| 02:03.48 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 02:04.06 | justinu | it's not difficult |
| 02:04.07 | *** join/#asterisk strib (n=ads@wompom.dur.ac.uk) |
| 02:04.49 | strib | Anyone here willing to help me configure Asterisk for voipuser.org? |
| 02:06.13 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 02:08.25 | fugitivo | Trazz: the asterisk samples come with nice examples for ivr, check the demo |
| 02:09.04 | strib | fugitivo: can you help me connect * to a sip gateway? |
| 02:09.24 | fugitivo | i can help you to solve your problems |
| 02:09.24 | Trazzz | what are some good companies to look at for inbound voip lines for my asterisk box? |
| 02:09.35 | fugitivo | strib: what's the problem? |
| 02:09.57 | strib | fugitivo: I set it up in sip.conf |
| 02:10.10 | strib | ..but when I reload it, it tells me it failed to register. |
| 02:10.21 | strib | I'll grep out the config lines. |
| 02:12.43 | riddlebox | Trazzz:I use broadvoice |
| 02:13.15 | Trazzz | riddlebox, how are they priced and what kind of features? |
| 02:13.19 | *** join/#asterisk netcrash (n=andre@82.102.38.99) |
| 02:13.47 | riddlebox | www.broadvoice.com can do a better job at telling you than me :) |
| 02:13.58 | justinu | try asterlink too |
| 02:14.33 | fugitivo | voicepulse, teliax |
| 02:15.54 | Trazzz | how do these companies deal with caller id inbound and outbound? |
| 02:16.27 | riddlebox | what the heck is going on, I had EXEC PLAYBACK working, now it doesnt |
| 02:16.49 | wunderkin | incoming caller id isnt working for me at the moment with broadvoice, they must be going through some changes again, someone on the list posted that they are being forced to change their 800 number |
| 02:17.56 | wunderkin | broadvoice wont send caller id |
| 02:18.03 | wunderkin | only the number on the account |
| 02:18.03 | Trazzz | hmm is broadvoice giving you inbound and outbound? |
| 02:18.06 | riddlebox | my incoming caller id works, but one time it will show you the town the call is coming from next time it will tell you the callers name |
| 02:18.08 | _Sam-- | i get caller id fine from teliax |
| 02:18.12 | *** join/#asterisk svenl_ (n=sven@AStrasbourg-251-1-3-6.w82-126.abo.wanadoo.fr) |
| 02:18.16 | _Sam-- | name and number |
| 02:18.25 | Trazzz | let me check out teliax |
| 02:18.30 | strib | Gah, laptop died |
| 02:18.41 | wunderkin | riddlebox, incoming caller id has always said unknown for me the past few days |
| 02:18.50 | *** join/#asterisk oej (n=oej@208.255.6.195) |
| 02:18.56 | strib | http://pastebin.com/509205 has the sip.conf bits, plus the diagnostics |
| 02:19.08 | _Sam-- | teliax is ok in my book, some reliability problems |
| 02:19.10 | riddlebox | really, maybe I do not get as many calls as you do |
| 02:19.12 | _Sam-- | but all in all not too bad |
| 02:19.12 | riddlebox | :) |
| 02:19.26 | _Sam-- | they need an east coast servre |
| 02:20.06 | Trazzz | they look pricey |
| 02:20.21 | riddlebox | i need to port a local number to broadvoice, everyone freaks out when I give them a chicago number, which is 5 hours away ;) |
| 02:20.39 | justinu | if you get asterlink, you can just bug file for help :P |
| 02:21.03 | wunderkin | i didnt have very good luck with asterlink, they use realtime :/ |
| 02:21.54 | Trazzz | so broadvoice is unlimited in state for 9.95 .. does that mean outbound? |
| 02:22.04 | Trazzz | will that be unlimited inbound from no matter where? |
| 02:22.20 | riddlebox | Trazzz:yes |
| 02:22.26 | Trazzz | perfect.. |
| 02:22.34 | Trazzz | i will use mci for outbound |
| 02:22.37 | riddlebox | and inbound as well |
| 02:22.37 | Trazzz | unlimited |
| 02:22.38 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:22.42 | mog_home | asterisk-dev |
| 02:22.52 | Qwell | mog_home: /j |
| 02:22.54 | Qwell | <PROTECTED> |
| 02:22.57 | strib | fugitivo: did http://pastebin.com/509205 give you enough information? |
| 02:23.03 | wunderkin | xxxpr0n |
| 02:23.07 | mog_home | thathanks |
| 02:23.10 | justinu | lol |
| 02:23.15 | Qwell | drunk much? |
| 02:25.13 | *** join/#asterisk psi_force (n=mark@c210-49-171-252.mckinn1.vic.optusnet.com.au) |
| 02:25.19 | *** join/#asterisk FelipeXxxx (n=FelipeX@201-221-196-3.bk11-dsl.surnet.cl) |
| 02:25.33 | psi_force | hi all |
| 02:28.00 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-253.msy.bellsouth.net) |
| 02:28.25 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 02:29.06 | *** join/#asterisk oej (n=oej@208.255.6.195) |
| 02:29.45 | psi_force | I'm having a problem with voicemail. For normal extensions voicemail works fine. However, I have created another voicemail entry not attached to any account. When a person gets diverted to the unavalible part of the server, they hear "extension XXX is not avaliable. please leave a message after the tone" but then the system hangs up on the user... without any recording taking place.. |
| 02:31.07 | *** join/#asterisk jmcc (n=jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 02:31.10 | _Sam-- | what does the console say |
| 02:31.30 | *** join/#asterisk iq (n=iq@ip24-252-63-164.om.om.cox.net) |
| 02:31.58 | _Sam-- | if i have a host that has a lot of deviation in the ping times to its IAX provider...will the jitterbuffer help that? |
| 02:31.59 | jmcc | dump question--does AEL support file #includes ? I can't seem to get it to work |
| 02:32.04 | Qwell | jmcc: no |
| 02:32.07 | Qwell | not currently |
| 02:32.26 | *** join/#asterisk Soul (n=Soul@87-196-39-131.net.novis.pt) |
| 02:32.38 | jmcc | Qwell: ok tnx, i thought I was going nuts |
| 02:32.54 | *** part/#asterisk iq (n=iq@ip24-252-63-164.om.om.cox.net) |
| 02:33.12 | jmcc | Qwell: but at least I've completely converted 27 contexts from extensions.conf to extensions.ael |
| 02:33.27 | Qwell | jmcc: cool |
| 02:33.38 | *** join/#asterisk droops (n=droops@pcp0010266914pcs.indpnd01.mo.comcast.net) |
| 02:33.49 | *** join/#asterisk Legend (n=legend@24.244.142.133) |
| 02:35.08 | Trazzz | what ports do i need open on the remote end user firewall to allow him to use softphone and for me to call him? |
| 02:35.36 | jpablo | <PROTECTED> |
| 02:36.39 | psi_force | do voicemail numbers (extensons) need to be numeric only or can they be words? |
| 02:37.28 | strib | psi_force: by default, the extension for a voicemail box is 'u' prepended to the standard extension. |
| 02:38.31 | psi_force | strib: cool but can I create a vociemail extension Sales and address it as uSales |
| 02:38.49 | strib | afaik, yes |
| 02:39.21 | ruud_org | kinda hard to select it in VoiceMailMain, though |
| 02:39.49 | _DAW-LAPTOP | yea, how would you dial letters? |
| 02:41.09 | strib | If you had the right context, you could dial direct. |
| 02:42.26 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 02:45.39 | Essobi | Packed in here tonight.. |
| 02:48.24 | *** part/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 02:48.28 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 02:48.42 | Essobi | Gee... Everyone don't talk at once. |
| 02:48.44 | strib | fugitivo: 'tis okay.. pebkac |
| 02:48.58 | droops | yeah its packed in here |
| 02:49.37 | droops | ive only been here once before, im taking an asterisk class this week |
| 02:50.09 | Trazzz | i am getting 499 not acceptable here on my sip phone |
| 02:50.11 | Trazzz | any ideas ? |
| 02:50.24 | justinu | hang out here and solve people's problems |
| 02:50.28 | justinu | there is no better class :P |
| 02:50.43 | fugitivo | ? |
| 02:50.51 | fugitivo | strib: ? |
| 02:51.04 | strib | problem exists between keyboard and chair |
| 02:51.15 | strib | *blush* I got the password wrong |
| 02:51.18 | wunderkin | hy me no astrix aix register sucky sucky five dollah |
| 02:51.37 | fugitivo | did i miss something? |
| 02:52.11 | strib | fugitivo: yeah, I sent you http://pastebin.com/509205 |
| 02:52.25 | Trazzz | wunderkin, i am now trying to connect up an xlite that is on the internet behind a firewall and he cant dial out with out not available here. i can dial him but no voice communications |
| 02:52.26 | Essobi | wunderkin rofl |
| 02:52.27 | fugitivo | oh |
| 02:52.29 | fugitivo | sorry |
| 02:52.49 | strib | Anyway, now I've registered, I *still* can't get a voice call from PSTN to my * box |
| 02:52.52 | strib | np |
| 02:54.15 | Essobi | Trazzz 499 is a client error |
| 02:54.36 | Essobi | Something's wrong with what you're dialing |
| 02:54.54 | Essobi | cause the server thinks there's something wrong w/it |
| 02:55.21 | Trazzz | this is remote client with xlite |
| 02:55.29 | Trazzz | i can call him but we hear not voices |
| 02:55.39 | Trazzz | he cant call out |
| 02:55.58 | _DAW-LAPTOP | Trazzz - you have NAT on both ends? |
| 02:56.03 | Trazzz | yes |
| 02:56.13 | _DAW-LAPTOP | Sip and nat are not friendly |
| 02:56.57 | Trazzz | i have asterisk box here local with xlite local and then xlite remote over internet |
| 02:56.59 | Trazzz | both use nat |
| 02:57.02 | Trazzz | can this work? |
| 02:57.09 | *** join/#asterisk Cyon (n=cyon@cyons.net) |
| 02:57.50 | Essobi | Trazzz Depends on your router/firewalls performing the nat.. |
| 02:57.58 | _DAW-LAPTOP | http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 02:58.08 | Essobi | if you got nat aware devies, or if your call manager is nat aware then you're fine. |
| 02:58.12 | justinu | hey cyon |
| 02:58.17 | Cyon | Hey justinu |
| 02:58.24 | Trazzz | i have nat aware firewalls for sure |
| 02:58.26 | _DAW-LAPTOP | they explain it better than me :) |
| 02:58.33 | Cyon | Anything going on? |
| 02:58.41 | justinu | nope |
| 02:58.48 | Cyon | Ah well. lol |
| 02:58.55 | Essobi | The question is are they nat SIP aware.. IE layer 7 markups for the corresponding public IP addresses.. |
| 02:59.28 | _DAW-LAPTOP | Otherwise oneway or noway voice |
| 02:59.32 | Essobi | if they're nat aware and session tracking the sip/rtp then you're fine.. |
| 02:59.39 | Essobi | _DAW-LAPTOP yup |
| 02:59.43 | Essobi | one way or no way.. |
| 03:01.14 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 03:01.19 | Darwin35 | hey kids |
| 03:01.25 | Darwin35 | an 16 19:56:05 NOTICE[5259]: res_odbc.c:294 load_odbc_config: registered database handle 'asterisk' dsn->[asterisk] |
| 03:01.25 | Darwin35 | Jan 16 19:56:05 NOTICE[5259]: res_odbc.c:552 odbc_obj_connect: Connecting asterisk |
| 03:01.25 | Darwin35 | Jan 16 19:56:05 WARNING[5259]: res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=2002 [MySQL][ODBC 3.51 Driver]Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) |
| 03:01.43 | Darwin35 | but it does not explain why |
| 03:01.46 | strib | Is your MySQL server running? |
| 03:01.50 | *** join/#asterisk minsu (n=minsu@59.12.193.2) |
| 03:01.51 | Darwin35 | yes |
| 03:02.09 | Qwell | and it's listening on that socket? |
| 03:02.10 | strib | 'mysql -h localhost root' work |
| 03:02.10 | Darwin35 | yes |
| 03:02.10 | strib | ? |
| 03:02.24 | Darwin35 | its on a diff box |
| 03:02.35 | Qwell | then...how are you going to have a unix socket in /tmp? |
| 03:03.08 | Qwell | So, the correct answer was "no, that isn't correct Qwell, the socket doesn't exist." :P |
| 03:03.15 | Katty | hmm. |
| 03:04.04 | *** join/#asterisk pengyong (n=lala@222.188.137.148) |
| 03:04.35 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 03:05.40 | minsu | whats vender use asterisk? |
| 03:05.53 | Essobi | Darwin35 umm. server isn't running, socket doesn't exist there.. uhh.. possibly permissions.. |
| 03:05.58 | *** join/#asterisk jef_ (i=fischer@p54846DAC.dip.t-dialin.net) |
| 03:06.00 | Essobi | minsu what what? |
| 03:06.20 | Essobi | "it's on a different box" |
| 03:06.43 | Essobi | then how do you expect to use the local socket to connect? |
| 03:06.48 | Qwell | Essobi: see above |
| 03:06.52 | minsu | Essobi, like Cisco . IP-PBX box. |
| 03:06.55 | Qwell | I already established this :P |
| 03:07.00 | Cyon | Qwell already settked that one... |
| 03:07.09 | Cyon | s/settked/settled |
| 03:07.16 | Essobi | minsu let me guess.. english isn't your first language? |
| 03:07.21 | Essobi | What're you trying to ask? |
| 03:07.23 | Darwin35 | sorry on phone |
| 03:07.25 | justinu | lol |
| 03:07.30 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:07.45 | Trazzz | ok now have calls going back and forth but we cant hear one another |
| 03:07.50 | Darwin35 | ok its using odbc libiodbc |
| 03:07.56 | Trazzz | anything special i have to do on the xlite configs? |
| 03:07.57 | minsu | Essobi, no, i'm not native speaker. sorry for my bad english. |
| 03:08.17 | Essobi | Trazzz whip out tethereal and look at the SDPs on the invites leaving the network.. |
| 03:08.23 | Essobi | can you sniff outside the router? |
| 03:08.37 | Essobi | is the * box on either of those local networks? |
| 03:08.43 | justinu | Essobi: i like they way you think |
| 03:08.46 | Trazzz | yes |
| 03:09.26 | Essobi | Trazzz try setting the reinvite=no and reloading the sip peers |
| 03:09.44 | Essobi | justinu I been cracking at this VoIP thing for years.. |
| 03:09.47 | Katty | hmm. |
| 03:09.54 | Darwin35 | I got around this once but cant remember how |
| 03:09.55 | justinu | essobi: only a few months for me |
| 03:10.12 | Qwell | Darwin35: by changing it from using the socket, to using a hostname? |
| 03:10.14 | *** join/#asterisk Trazz (n=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 03:10.25 | Trazz | esobi, i am local to the * box |
| 03:10.32 | Essobi | trazz: PRoblems? |
| 03:10.34 | Essobi | hehe |
| 03:10.46 | Trazz | ya someting on the inet connectin |
| 03:10.51 | Essobi | I noticed |
| 03:11.06 | Essobi | Trazzz try setting the reinvite=no for each peer and reloading the sip peers |
| 03:11.13 | Essobi | and reconnecting both clients |
| 03:11.15 | Trazz | ok one sec |
| 03:11.16 | Essobi | and make a call |
| 03:11.26 | Essobi | that might keep rtp local to the box |
| 03:11.47 | Essobi | but if the * box is behind the firewall too.. mmmm.. might not like it.. |
| 03:11.49 | Netgeeks | katty had your warm soy milk yet today/ |
| 03:11.53 | justinu | Essobi: don't recall seeing you here before |
| 03:12.07 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 03:12.09 | Katty | Netgeeks: i think i switched to rice milk. |
| 03:12.17 | fugitivo | Katty: you shouldn't say "hmm." until tomorrow morning |
| 03:12.28 | Netgeeks | ack, no rice milk here ;9 |
| 03:13.14 | Katty | fugitivo: that does not parse. |
| 03:14.07 | Netgeeks | oops, no shift key either.... curses Synergy you evil remote keyboard shift losing app! |
| 03:14.12 | Netgeeks | :( |
| 03:14.16 | Darwin35 | so change the res_mysql.conf dbsocket |
| 03:14.17 | justinu | lol |
| 03:14.39 | Trazz | essobi, tried it.. no voice still |
| 03:14.42 | Qwell | no, remove the socket, and make it use a hostname |
| 03:14.44 | justinu | netgeeks: you're a funny guy, but we need to introduce you to a real life woman ;) |
| 03:14.59 | Netgeeks | two divorces say otherwise |
| 03:15.04 | justinu | ack |
| 03:15.57 | fugitivo | two divorces? |
| 03:16.16 | Katty | Netgeeks: you'll have better luck with a kitty. |
| 03:16.20 | Katty | Netgeeks: trust me. |
| 03:16.22 | Netgeeks | yep, you'd think the first one might have taught me something |
| 03:16.37 | Qwell | Netgeeks: 8th time's a charm |
| 03:16.41 | fugitivo | Netgeeks: they left you? |
| 03:16.42 | Darwin35 | so socket=hostname ? |
| 03:16.48 | Netgeeks | I've got two kitties... I don't plan on marrying either |
| 03:16.49 | Qwell | no |
| 03:16.50 | Qwell | hostname= |
| 03:16.54 | Qwell | remove the socket crap entirely |
| 03:17.06 | Darwin35 | ok |
| 03:17.24 | Netgeeks | fug: first one I left (whew), second was a mutual go our own way thing |
| 03:17.39 | pifiu | wasup everyone |
| 03:17.40 | justinu | A U.S. F-15 plane has crashed near Okinawa island, Japan, a Japanese Defense official says. |
| 03:18.07 | *** join/#asterisk FastJack_ (i=fastjack@p5091CAF2.dip.t-dialin.net) |
| 03:18.20 | fugitivo | what was doing a US F15 near Okinawa? |
| 03:18.38 | justinu | i think we still use japan as a military base |
| 03:18.42 | Netgeeks | Katty: yep, kitties don't try to make me grow up or change... so we'll be fine me and my two cats |
| 03:18.52 | *** join/#asterisk Snooker (n=klayton@ras-27.expert.com.br) |
| 03:19.37 | fugitivo | a dog is better |
| 03:19.47 | Katty | fugitivo: dogs smell. |
| 03:19.54 | justinu | lol |
| 03:19.57 | justinu | that's true |
| 03:20.06 | Katty | fugitivo: and drool. |
| 03:20.11 | fugitivo | not if you clean it |
| 03:20.14 | Katty | fugitivo: and dig and bark |
| 03:20.16 | justinu | cats rules!! |
| 03:20.22 | Katty | annoying pesky creatures if you ask me |
| 03:20.23 | Netgeeks | nah, I'm not much for a dog, had them the whole time I was growing up... you need to pay too much attention to them, and they wake you up when you want to sleep in so they can go outside |
| 03:20.25 | [av]bani | fugitivo: this might have a little tiny something to do with it -> http://www.kadena.af.mil/ |
| 03:20.26 | fugitivo | Katty: that's why dogs are better |
| 03:20.34 | [av]bani | fugitivo: but just only a tiny little bit. |
| 03:20.40 | fugitivo | a cat is boring |
| 03:20.47 | justinu | dogs have seperation anxiety |
| 03:20.49 | Darwin35 | its in only 1 file |
| 03:20.54 | fugitivo | you can train a dog |
| 03:20.56 | Darwin35 | and its still the same error |
| 03:21.20 | fugitivo | mine is a border collie, smart dog |
| 03:21.28 | justinu | collies are cute |
| 03:21.32 | Netgeeks | thats what I had growing up, very smart dog |
| 03:21.40 | Netgeeks | but, no more dogs for me |
| 03:22.06 | fugitivo | you have to see mine while jumping |
| 03:23.13 | fugitivo | i'll teach him to catch the frisbee :) |
| 03:24.10 | justinu | that long fur must be a bitch to keep clean tho |
| 03:24.30 | pifiu | does broadvoice support IAX? or SIP? |
| 03:24.53 | Netgeeks | I was thinking I should place a bid on one of the jobs on totaro's helpdesk thingy, and say that the ability to speak ancient sumerian is the key skill he should be judging me on |
| 03:26.36 | Netgeeks | I can't say for sure, never used them, pifiu, but I'm pretty confident they do not support iax |
| 03:28.08 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-249.dsl.ltrkar.swbell.net) |
| 03:28.09 | pifiu | ok now i know you guys are going to kill me for the next questions |
| 03:28.15 | Qwell | pifiu: probably |
| 03:28.21 | Qwell | only one way to find out |
| 03:28.34 | watchy | is it possible to do all asterisk configs through sql? |
| 03:28.37 | pifiu | but last time i used asterisk someone else set it up for me, and i just learned it from there, messing with the files and such |
| 03:28.47 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 03:28.54 | wunderkin | pifiu: broadvoice is sip |
| 03:28.57 | pifiu | but i have never installed asterisk sucessfully for myself without running into problems |
| 03:29.06 | Netgeeks | watchy, you would need at least one config file to tell asterisk where to look for the others... |
| 03:29.16 | watchy | Netgeeks: of course |
| 03:29.18 | pifiu | is there any way someone can lead me to a step by step or help me out? i know its not hard, just want someone by my side while i do it |
| 03:29.31 | watchy | do most people just do cdr/peers in sql? |
| 03:29.50 | Netgeeks | mind if I share that popcorn, Cyon? this could be interesting |
| 03:30.06 | pifiu | lol |
| 03:30.09 | pifiu | ok so i guess no takers? |
| 03:30.20 | Qwell | pifiu: $ |
| 03:30.26 | pifiu | lol |
| 03:30.33 | pifiu | 1 dolla! |
| 03:30.41 | justinu | i'd buy that for a dollar |
| 03:30.45 | Qwell | Qwell needs beer money for ETel :P |
| 03:30.52 | fugitivo | paypal paypal paypal |
| 03:31.04 | Netgeeks | pifiu, have you tried installing Asterisk@Home? |
| 03:31.10 | pifiu | lmao funny how the $ comes up and plenty of people reply |
| 03:31.10 | Qwell | Netgeeks: HEATHEN! |
| 03:31.12 | pifiu | yes i ahve |
| 03:31.18 | fugitivo | Netgeeks: NO! |
| 03:31.29 | pifiu | really i just wanted the experience of doing htis from the source |
| 03:31.38 | Qwell | pifiu: make && make install |
| 03:31.40 | Qwell | simple |
| 03:31.51 | Netgeeks | fug: hey AAH is a good place for someone who can't get the source to install to start |
| 03:31.59 | fugitivo | pifiu: this channel is for troubleshooting, if you want someone to teach you, use paypal :) |
| 03:32.02 | Netgeeks | sure, it may ruin them for life |
| 03:32.11 | Netgeeks | but they'll be making phone calls |
| 03:32.20 | fugitivo | Netgeeks: no, it's a bad place because it doesn't teach you anything |
| 03:32.21 | pifiu | well netgeeks, i know how to somewhat use asterisk |
| 03:32.32 | pifiu | just want to get more into it and start messing with it from scartch |
| 03:32.44 | Netgeeks | installing it is the easy part actually |
| 03:32.54 | *** join/#asterisk Weezey (n=ohno@206.210.109.232) |
| 03:33.19 | Weezey | anyone know if it's possible to change the Cisco soft menus on a 7940 from the telnet? |
| 03:33.31 | Darwin35 | still no go |
| 03:33.35 | Darwin35 | this sucks |
| 03:33.45 | Legend | Weezey: the softkeys? afaik you cannot change them on a SIP load |
| 03:34.12 | Netgeeks | pifiu I can give you some basic directions, but they won't differ from the wiki, and they will in fact be less lucid |
| 03:34.37 | pifiu | ok tell you what im going to try |
| 03:34.41 | Weezey | I mean the options menu. |
| 03:34.44 | pifiu | i just hope i dont break anything |
| 03:34.44 | Netgeeks | but I can't really spend the time to teach you, I took a short break from my coding madness to ... take a break... |
| 03:34.45 | pifiu | =P |
| 03:34.56 | Netgeeks | you have the source code? |
| 03:35.03 | Weezey | I made the TFTP manual from the phone itself and I want to change it back to the dhcp one. |
| 03:35.08 | pifiu | ok well i wasnt expecting coding or anyhting just from the download to the compiling |
| 03:35.11 | Weezey | but I'm not at the phone. |
| 03:35.14 | Netgeeks | zaptel, libpri, and asterisk specifically, also asterisk-addons if you plan to do some SQL stuff |
| 03:35.17 | pifiu | the sip and iax and any other registration i can do |
| 03:35.51 | Legend | Weezey: ah, never tried, sounds possible though |
| 03:36.08 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
| 03:36.32 | Weezey | Legend: I would think it should be. |
| 03:36.41 | Legend | but doesn't look like it |
| 03:36.56 | Weezey | My innitial read on voip info said that you can't make a line auto-answer unless you're at the phone |
| 03:37.04 | Weezey | so I'm wondering if it's not possible. |
| 03:37.33 | Netgeeks | pifiu: first thing is grab the source, I think 1.2.2 was recently relesed |
| 03:37.37 | Netgeeks | released |
| 03:37.55 | *** join/#asterisk copantl (n=copantl@205.240.200.113) |
| 03:38.37 | justinu | why is it that all the israeli PM's who try and make peace with the palestinians get taken out somehow? |
| 03:39.48 | Netgeeks | *sidestep* |
| 03:40.24 | justinu | just wondering out loud |
| 03:40.56 | Cyon | Netgeeks: No, mine! All mine! *holds out the bowl* lol |
| 03:40.57 | copantl | any body knows howto place a CID in oh323.conf for billing purpose? |
| 03:41.21 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 03:41.22 | Netgeeks | I'm not sure how CID coorelates with billing? |
| 03:41.56 | Netgeeks | by CID, I take it you mean caller id? |
| 03:42.06 | copantl | yes |
| 03:42.10 | copantl | caller id |
| 03:42.18 | Netgeeks | and you are doing your billing based upon the caller id? |
| 03:42.23 | pifiu | netgeeks let me ask you |
| 03:42.31 | copantl | yes |
| 03:43.14 | pifiu | i remember compiling a long time ago the whole zaptel, libri and asterisk |
| 03:43.16 | Netgeeks | I'm not so sure that is a great idea, but I assume you've already gone past that decision.... |
| 03:43.19 | pifiu | do i just need asterisk now? |
| 03:43.39 | pifiu | i do have a modem for 911 dialing |
| 03:43.50 | Netgeeks | It's been a very long time since i did any oh323, and I don't remember any caller id directives in the config |
| 03:44.16 | Netgeeks | you could do some magic in the dialplan by capturing the channel and coorelating a caller id setting with the channel |
| 03:44.46 | copantl | oh323---------------oh323(needcallerid)-------(billingreceivecid)-----E1 |
| 03:44.51 | Netgeeks | pifiu: If I were you, I'd grab the source code (it all has to match versions) and compile it... you at a minimum want zaptel libpri and asterisk |
| 03:45.13 | *** join/#asterisk Legend` (n=legend@24.244.142.133) |
| 03:45.42 | Netgeeks | then you compile zaptel, librpi and asterisk in that order |
| 03:46.21 | Netgeeks | cd zaptel ; make ; make install ; cd ../libpri ; make ; make install ; cd ../asterisk ; make ; make install <-- assuming they are all in the same directory |
| 03:48.21 | pifiu | so grab source for asterisk, grab source for zaptel and then grap source for librpi |
| 03:48.25 | copantl | How can i do coorelating with dialplan.. whe i use amportal and a2billing? |
| 03:48.30 | pifiu | and compile each |
| 03:48.58 | Netgeeks | pifiu, yes in the order I mentioned aboce |
| 03:49.00 | Netgeeks | above |
| 03:49.04 | pifiu | ok thanks |
| 03:50.26 | Netgeeks | copantl: hrm, I would probably use Cut app with channel to get the unique h323 channel (like OH323/humptydumpty) from the variable CHANNEL and then use DBGET to pull a callerid that I had previously associated with that channel |
| 03:50.41 | Netgeeks | or.... |
| 03:50.42 | *** join/#asterisk bmg505 (n=leon@c1-135-16.rndf.isadsl.co.za) |
| 03:51.02 | Netgeeks | I might look into using gnugk and seeing if I could do something less hackish by using it |
| 03:51.04 | Corydon76-home | DBGet is deprecated |
| 03:51.08 | copantl | wow? |
| 03:51.32 | Netgeeks | death to the new way! |
| 03:51.47 | Corydon76-home | There's now a DB function |
| 03:52.07 | Corydon76-home | ${DB(family/key)} |
| 03:52.15 | Netgeeks | what he said |
| 03:52.42 | Corydon76-home | which you can set, as well: Set(DB(family/key)=value) |
| 03:53.34 | Corydon76-home | All sorts of neat things you can do with functions... |
| 03:53.37 | Netgeeks | wow? ? |
| 03:53.44 | pifiu | hey netgeeks what ever happedn to "zapata" |
| 03:53.45 | pifiu | ? |
| 03:53.47 | pifiu | is that gone? |
| 03:53.54 | Netgeeks | zapata has been deprecated |
| 03:53.58 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 03:53.59 | Netgeeks | ;) |
| 03:54.00 | Corydon76-home | Netgeeks: thanks for the sarcasm |
| 03:54.14 | pifiu | gotcha |
| 03:54.17 | *** join/#asterisk welles (n=welles@222.90.92.113) |
| 03:54.23 | pifiu | see i remember doing thihs |
| 03:54.28 | pifiu | now if i ever did it good or bad i dont know |
| 03:54.28 | pifiu | lol |
| 03:54.39 | elvisthedj | good <insert appropriate time of day here> all |
| 03:54.42 | copantl | is to high for a beginner like me |
| 03:54.52 | Corydon76-home | Netgeeks: I just think functions are hot shit because I wrote the implementation in 1.2 |
| 03:55.42 | Netgeeks | I'm way way out on a shakey limb here, but zapata had something to do with the tormenta/zapata cards and has been un-needed for quite some time. As for what it did when it was needed... I'm clueless |
| 03:56.23 | Corydon76-home | zapata was a library that did the tones for tonezones |
| 03:56.38 | Netgeeks | Corydon: don't mind me, I'm nto looking forward to converting my dialplan to the 'new' way... *sigh* so I get edgy when I think of the function changes and application deprecations |
| 03:56.39 | ast_new-B | how can prepaid card integrated with asterisk ? just like a2billing but it can operate in transparently |
| 03:56.49 | Lots | looks like iaxtel isn't working at the moment, anyone have any other iax providers i can try for free? |
| 03:56.55 | elvisthedj | i'm really kinda sad about db_put and db_get being gone ... i didn't count on having to rewrite my dialplan when i build cvs a few nights ago |
| 03:56.58 | Corydon76-home | The functionality of the zapata library has been incorporated directly into Asterisk |
| 03:56.59 | Netgeeks | Lots: FWD |
| 03:57.21 | Corydon76-home | elvisthedj: they're not gone, they're deprecated |
| 03:57.29 | Corydon76-home | elvisthedj: they'll be gone in 1.4, however |
| 03:57.49 | Netgeeks | they have been sent to live in an assisted living home, go visit them while you can |
| 03:57.54 | Corydon76-home | You have an entire release cycle to transition to the new DB function |
| 03:58.27 | elvisthedj | Corydon76-home: yeah, but i trashed all my apps and they didn't build w/ cvs.. unless it puts them somewhere else and i just need to copy them over (that'd be very gentooish if they did) |
| 03:59.09 | copantl | what is the difference between DNID and CID? |
| 03:59.10 | Netgeeks | I think they are gone in cvs |
| 03:59.12 | elvisthedj | Corydon76-home: i suppose i should go ahead and do this now anyway.. so i'm prepared.. i'm just not understanding how to use the DB "function" in the dialplan |
| 03:59.32 | Corydon76-home | DNID is the number dialled. CID is the number that is calling |
| 03:59.55 | Netgeeks | Actually CID is the number that has been presented as the number calling |
| 04:00.03 | elvisthedj | perhaps i should just start using mysql and quit using the asterisk db |
| 04:00.11 | Nugget | ewwww |
| 04:00.25 | Corydon76-home | Netgeeks: technically, DNID isn't the number dialled either |
| 04:00.39 | Corydon76-home | Netgeeks: but it's close enough to be understood |
| 04:01.08 | Netgeeks | elvis: depends on what you are doing. If you just need to store and retrieve static data from the asterisk dialplan I wouldn't choose to bring mysql and all the attendant trash required to use it into play |
| 04:01.10 | Lots | does FWD use IAX? |
| 04:01.38 | Corydon76-home | Or you could copy func_odbc into 1.2 and use that... |
| 04:02.00 | pifiu | netrgeeks do i haev to modprobe after zaptel? |
| 04:02.27 | Corydon76-home | zaptel is just the general driver, you need a specific driver after it |
| 04:02.35 | Netgeeks | if on the other hand you want to build a back-end DB thjat allows your asterisk dial plan and other outside sources to share data.... then an external DB might be your friend |
| 04:02.37 | elvisthedj | Netgeeks: Yeah, I'm a home user (not an @home user though) and use it for things like playing back names of callers, setting my caller id ... blah blah.. but when I try to use DB function, I get invalid app.. perhaps functions are meant for ael only?? |
| 04:02.44 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net) |
| 04:02.59 | Corydon76-home | elvisthedj: it's a function, not an app |
| 04:03.05 | Netgeeks | pifiu, do you have a digium or compatable PCI card in the system? |
| 04:03.07 | elvisthedj | Corydon76-home: tell asterisk that |
| 04:03.08 | elvisthedj | :) |
| 04:03.19 | Corydon76-home | elvisthedj: you don't call it like you call an app |
| 04:03.33 | Corydon76-home | elvisthedj: a function is more like a variable |
| 04:03.34 | pifiu | yes i do |
| 04:03.41 | Netgeeks | what card? |
| 04:03.42 | pifiu | the modem |
| 04:03.50 | Netgeeks | X100P? |
| 04:03.53 | pifiu | yes |
| 04:04.13 | pifiu | do i need to probe that now though? |
| 04:04.14 | elvisthedj | Corydon76-home: ahh.. i'm following the syntax on the wiki, but it doesn't really show it "in the dial plan" .. gives me no more info than Show function does from the cli |
| 04:04.25 | Netgeeks | um, uh, thats the wcfxo module, I think, long time since I loaded one of those |
| 04:04.25 | ast_new-B | can someone please tell me what is DISA mean ? |
| 04:04.30 | Qwell | ~disa |
| 04:04.34 | jbot | somebody said disa was direct inward system access. show application disa |
| 04:04.35 | Corydon76-home | elvisthedj: ${DB(family/key)} to retrieve the value |
| 04:04.45 | Qwell | DISA (Direct Inward System Access) |
| 04:04.47 | Qwell | nm |
| 04:04.50 | Netgeeks | damn jbot, you type fast |
| 04:04.50 | Corydon76-home | elvisthedj: Set(DB(family/key)=value) to set it |
| 04:04.51 | Qwell | how did I miss that? |
| 04:05.16 | ast_new-B | and how it operate ? |
| 04:05.17 | elvisthedj | Corydon76-home: Thanks. I'll go put that info to use right now ;) |
| 04:05.19 | *** join/#asterisk jbroome (n=jbroome@63-168-10-93.celito.net) |
| 04:05.49 | Netgeeks | if modprobe wcfxo doesn't work try wcfxs... I still can't ever get those straight |
| 04:06.24 | Corydon76-home | s/wcfxs/wctdm/ |
| 04:06.36 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 04:07.03 | pifiu | its the one where a landline connects to |
| 04:07.09 | pifiu | fxs i belive is the name |
| 04:07.12 | Corydon76-home | That's wcfxo |
| 04:07.16 | pifiu | oh |
| 04:07.17 | pifiu | so fxo |
| 04:07.27 | pifiu | its the stupid asterisk modem |
| 04:07.44 | Corydon76-home | x100p is driven with the wcfxo driver |
| 04:08.00 | Corydon76-home | TDM400P is driven with the wctdm driver |
| 04:08.01 | pifiu | ok so let me ask you |
| 04:08.10 | pifiu | after i make, make install the zaptel, should i probe that? |
| 04:08.14 | pifiu | or how should i do it? |
| 04:08.23 | Netgeeks | modprobe wcfxo |
| 04:08.31 | Corydon76-home | modprobe wcfxo will load zaptel as a dependency |
| 04:08.36 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 04:08.38 | pifiu | meaning? |
| 04:08.50 | Corydon76-home | A dependency? |
| 04:08.54 | justinu | it's automagic |
| 04:08.58 | Netgeeks | all you have to do is 'modprobe wcfxo' |
| 04:09.09 | pifiu | ok this is all before i do librpi? |
| 04:09.10 | Netgeeks | soon as you have successfully compiled and installed zaptel |
| 04:09.15 | Corydon76-home | It needs that driver, so it automatically loads it |
| 04:09.17 | pifiu | ok yes it compiled |
| 04:09.23 | Netgeeks | you don't have to do it to do the rest of the compiles |
| 04:09.26 | pifiu | so next thing probe and move on to librpi |
| 04:09.27 | justinu | libpri is a library, not a kernel module |
| 04:09.37 | Netgeeks | you only need to load it when you want to actually use the card |
| 04:09.40 | elvisthedj | Corydon76-home: so, before i go changing everything, does this priority look ok? exten => s,2,Set(SCREENFILE=${DB(callerid/${CALLERIDNUM})}) |
| 04:09.53 | Corydon76-home | Almost |
| 04:09.56 | pifiu | i might not use it for a month, but does it hurt if i load it now? |
| 04:09.56 | *** join/#asterisk Strom_C (n=strom@216-80-66-245.lem-bsr1.chi-lem.il.cable.rcn.com) |
| 04:10.02 | Netgeeks | no |
| 04:10.06 | Corydon76-home | CALLERIDNUM is about to be deprecated |
| 04:10.06 | Netgeeks | it doesn't hurt |
| 04:10.12 | elvisthedj | ah |
| 04:10.13 | Corydon76-home | Use ${CALLERID(num)} instead |
| 04:10.17 | pifiu | ok let me do that |
| 04:10.18 | elvisthedj | :D ok |
| 04:10.20 | Netgeeks | JOy of Joys |
| 04:10.21 | pifiu | sorry for so many questions |
| 04:10.34 | elvisthedj | Corydon76-home: but otherwise, that's ok |
| 04:10.39 | Netgeeks | I can't wait until I am forced to move to the new deprecation free code |
| 04:10.48 | Corydon76-home | The nice thing about the function for cid is that we don't need a special application anymore to set them |
| 04:11.01 | Netgeeks | *cry* |
| 04:11.23 | Corydon76-home | Oh, we'll continue to deprecate things, as we find better ways to do things |
| 04:11.32 | Netgeeks | I like my carburators! why do I have to buy a car with fuel injection, can't I just keep getting a carburatuer? |
| 04:11.43 | pifiu | hey im reading some random guide, it also says modprobe zaptel and then after that do the wcfxo |
| 04:11.46 | Netgeeks | too bad I can't spell them |
| 04:11.50 | elvisthedj | lol |
| 04:11.52 | pifiu | whats the point of probing zaptel? is that necessary? |
| 04:11.53 | Netgeeks | no |
| 04:11.58 | Netgeeks | don't worry about zaptel |
| 04:12.01 | pifiu | ok |
| 04:12.02 | *** join/#asterisk oej (n=oej@208.255.6.195) |
| 04:12.04 | pifiu | well its probed |
| 04:12.08 | Netgeeks | it will be automagiced |
| 04:12.12 | pifiu | wcfxo at least |
| 04:12.13 | pifiu | ok |
| 04:12.18 | copantl | any body use oh323? |
| 04:12.23 | pifiu | is that a made up word? |
| 04:12.28 | Netgeeks | quite |
| 04:12.33 | Essobi | copantl I did |
| 04:12.50 | Essobi | havn't for a while.. |
| 04:12.54 | *** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 04:13.07 | copantl | what's mean amaFlags=default? |
| 04:13.16 | Essobi | Netgeeks I think the proper termis automagically |
| 04:13.18 | copantl | in the oh323.conf |
| 04:13.18 | Essobi | ;) |
| 04:13.31 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:13.35 | silentfury | got a quick polycom 301 question.. has anyone seen them refuse to sync their time even though they have a proper ntp server set? |
| 04:13.48 | Netgeeks | but I wanted a past tense verb, and not a verb modifier! |
| 04:13.54 | Netgeeks | so I made one up! |
| 04:14.12 | copantl | there are the posible options :AMA flags (default, omit, billing, documentation) |
| 04:14.29 | copantl | sorry for my english :( |
| 04:14.42 | Netgeeks | your english appears to be better than mine |
| 04:15.16 | pifiu | ok its making asterisk |
| 04:15.23 | Essobi | copantl what about it? |
| 04:15.32 | Essobi | silentfury http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&p=21695 |
| 04:15.41 | pifiu | yes silent |
| 04:15.48 | pifiu | they take like 5 minutes to sync |
| 04:15.54 | pifiu | for some odd reason |
| 04:15.56 | Essobi | youch |
| 04:16.00 | pifiu | im running 501's |
| 04:16.02 | copantl | i just wanna know howto bill a oh323 channel |
| 04:16.15 | pifiu | and at the beginning when they get restarted it take them a couple of minutes to sync up |
| 04:16.32 | silentfury | these phones don't sync. ever. |
| 04:17.12 | copantl | because i receive 30 calls from this channel but i don't know how to bill |
| 04:17.22 | *** part/#asterisk minsu (n=minsu@59.12.193.2) |
| 04:18.17 | copantl | i'm trieing using CID but i don't know where place a CID on oh323.conf |
| 04:18.19 | Lots | is there anyone on FWD i can call to check my connection? |
| 04:18.32 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 04:18.40 | Netgeeks | copantl, you are going to need someone who is familiar with your billing application and your asterisk configuration to look at that. I doubt very much, you'll be able to resolve your issue here unless you come across someone who is trying to do the exact same thing you are |
| 04:18.41 | pifiu | hmmm silent |
| 04:18.49 | silentfury | is there any debug mode to check wtf is happening with the polycom's and the ntp sync? |
| 04:18.51 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 04:18.57 | pifiu | try to update to the latest firmware? |
| 04:19.08 | Netgeeks | silentfury sniff the line and look at the packets |
| 04:19.13 | justinu | silentfury: probably... do you hae the admin guide? |
| 04:19.16 | pifiu | silent, im no expert on polycom, i use them, but im no master |
| 04:19.19 | pifiu | learning about them still |
| 04:19.34 | copantl | ok |
| 04:19.34 | silentfury | justin: the polycom admin guide? |
| 04:19.40 | justinu | yeah |
| 04:19.46 | silentfury | nope.. |
| 04:19.50 | justinu | you need that |
| 04:20.01 | pifiu | contact the place where you bought ti from |
| 04:20.03 | justinu | there's a setting in sip.cfg you can modify to increase the polycom log verbosity. |
| 04:20.17 | pifiu | he left |
| 04:20.28 | justinu | oh |
| 04:20.31 | justinu | dork |
| 04:20.56 | Netgeeks | maybe he had to walk his dog |
| 04:21.07 | Utah_Dave | I'm having trouble with my iax.conf to let my iax softphone (iaxComm) connect to my * box. Anyone got a second to help out? |
| 04:21.48 | Netgeeks | I have just one question... why would you want to use an iax softphone? |
| 04:21.49 | pifiu | should I make progdocs? |
| 04:21.52 | *** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 04:21.56 | silentfury | sigh |
| 04:22.04 | silentfury | sorry about that |
| 04:22.09 | Corydon76-home | pifiu: are you a programmer? |
| 04:22.12 | silentfury | justin: Can you give me the link for the admin guide |
| 04:22.18 | pifiu | no i am not |
| 04:22.21 | Utah_Dave | Netgeeks: It seems easier to use with nat than sip |
| 04:22.25 | pifiu | but im confused if they are guides? |
| 04:22.33 | Corydon76-home | pifiu: why would you need programming documentation, then? |
| 04:22.36 | Netgeeks | Utah_Dave, ah, okay, thanks |
| 04:22.40 | pifiu | says progdocs |
| 04:22.46 | pifiu | thought it meant program documents |
| 04:22.51 | Essobi | copantl the AMA flags in h323 are just for disposition of the call based on the channel |
| 04:23.01 | Essobi | copantl http://www.voip-info.org/wiki/view/Asterisk+billing |
| 04:23.16 | pifiu | cory so in short dont install it |
| 04:23.24 | Netgeeks | pifiu: correct |
| 04:23.25 | *** join/#asterisk sryan2k1 (n=spencer@66-227-158-114.dhcp.bycy.mi.charter.com) |
| 04:23.27 | pifiu | ok |
| 04:23.28 | *** join/#asterisk JoJoStomp (n=JoJoStom@adsl-065-012-166-198.sip.asm.bellsouth.net) |
| 04:23.32 | Corydon76-home | You may, but I doubt it'll provide any use to you |
| 04:23.33 | pifiu | so i have sucessfully installed asterisk! |
| 04:23.54 | pifiu | is it set to run at startup? |
| 04:23.59 | Corydon76-home | Nope |
| 04:24.01 | Netgeeks | you did make samples? |
| 04:24.07 | silentfury | all i know is |
| 04:24.08 | Corydon76-home | You need to configure it first |
| 04:24.08 | pifiu | yes i made samples |
| 04:24.11 | silentfury | i am NEVER choosing polycom phones again. |
| 04:24.18 | copantl | thanks Essobi |
| 04:24.19 | Netgeeks | okay, now you get to configure it |
| 04:24.20 | pifiu | silent why? wtf those phones awes eome |
| 04:24.31 | Corydon76-home | pifiu: which distro? |
| 04:24.36 | pifiu | ok before i set it to startup on startups i should configure it? |
| 04:24.38 | pifiu | fedora 4 |
| 04:24.44 | Lots | is there an option somewhere in the FWD.communicator i tell it to use IAX2 and route the calls through my asterisk server? |
| 04:24.57 | Corydon76-home | You could probably use the redhat startup script, then, in contrib/init.d/ |
| 04:25.12 | Netgeeks | pifiu, yes, I would think at a minimum you would want to configure your x100p to work |
| 04:25.29 | Netgeeks | however if you want to make sure you installed it |
| 04:25.30 | pifiu | wont use it for another couple of week |
| 04:25.39 | Netgeeks | you should be able to fire up asterisk and see if it runs |
| 04:25.48 | sryan2k1 | Question, is asterisk able to call a number, and wait for a DTMF response from that number? for an example, you call extension 3, i want the asterisk system to call 1-734-111-1111, and when the other end answers, listen for a DTMF tone, if there is one sent back, connect the two calls, otherwise do something else with it |
| 04:25.54 | Netgeeks | you could run it in console for verification |
| 04:26.04 | Netgeeks | asterisk -vvvgc |
| 04:26.07 | Netgeeks | should do it |
| 04:26.28 | pifiu | says ready! |
| 04:26.29 | pifiu | wohoo |
| 04:26.31 | Netgeeks | I think asterisk will come up clean on the sample files |
| 04:26.35 | mog_home | sure |
| 04:27.05 | pifiu | wow the command exit on the cli doesnt work? |
| 04:27.11 | justinu | stop now |
| 04:27.17 | pifiu | oh |
| 04:27.19 | Netgeeks | not when you are in console... |
| 04:27.31 | Netgeeks | exit works when you use asterisk -r to connect to an already running asterisk |
| 04:27.36 | pifiu | in my other machine i haev it tset to startup |
| 04:27.39 | *** part/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 04:27.40 | pifiu | and i alwys just do asterisk -r |
| 04:27.44 | Netgeeks | you are running asterisk in your terminal console right now |
| 04:27.48 | pifiu | oh |
| 04:28.18 | pifiu | ok this is neat |
| 04:28.26 | pifiu | netgeeks i appreciate your help and everyone elses |
| 04:28.30 | pifiu | see it wasnt that bad! |
| 04:28.30 | Netgeeks | have fun! |
| 04:28.46 | pifiu | ill be around here though, but interesting to always learn something |
| 04:28.47 | Netgeeks | when they come to take you away in the white rubber van, don't blame me |
| 04:28.50 | wunderkin | any sphinx3 helpers? |
| 04:28.52 | pifiu | lol |
| 04:28.55 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
| 04:29.03 | *** part/#asterisk oej (n=oej@208.255.6.195) |
| 04:29.23 | sryan2k1 | can someone recomend a good VoIP provider in the united states, cheap is good, but reliability is more important (for a business) |
| 04:29.50 | *** join/#asterisk brookshire[home] (n=matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 04:30.35 | elvisthedj | ok, so no more setcidnum either? i use SetCallerId the same way? |
| 04:31.08 | Netgeeks | I think all the Set apps have beenn functionized |
| 04:31.35 | BoRiS | Anyone know how to put a default configuration in a sipura (2100) firmware image and if they try to factory reset the device, the default configuration takes presidence? |
| 04:31.37 | elvisthedj | sryan2k1: there is a good list on the wiki at voip-info.org .. depends on your location, how many channels, sip, iax.. |
| 04:32.06 | wunderkin | mog_home did you say you were a sphinx boy? :P |
| 04:32.12 | elvisthedj | Netgeeks: well, i see the SetCallerID app.. I've never used it, so i didn't know if it was new, but the console help doesn't say deprecated |
| 04:32.18 | sryan2k1 | would you recomend SIP or IAX |
| 04:32.19 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 04:32.36 | Netgeeks | I'm the wrong person to ask, I am not using any of the new function stuff |
| 04:32.47 | elvisthedj | syle: I like IAX.. cuz it's native to asterisk and easier on firewalls |
| 04:33.01 | elvisthedj | oops.. i mean sryan2k1 :) |
| 04:33.03 | Netgeeks | I've got about 10,000 lines of dialplan code that I'm not currently interested in having to bring up to date |
| 04:33.10 | sryan2k1 | :), ok thanks ill take a look |
| 04:33.35 | elvisthedj | sryan2k1: I use both teliax and junctionnetworks for iax, but i'm not a business, so you might need more channels than me |
| 04:33.46 | elvisthedj | sryan2k1: but i've had no problems with either |
| 04:34.09 | Netgeeks | teliax charges for minutes inbound as well as outbound tho |
| 04:34.18 | sryan2k1 | how many channels do you have? |
| 04:35.30 | elvisthedj | sryan2k1: I've never used more than 2 at once .. per provider |
| 04:35.31 | Lots | FWD communicator how the heck to i get it to connect through my asterisk server? |
| 04:35.55 | justinu | lots: i think it's locked to fwd |
| 04:35.58 | elvisthedj | sryan2k1: i don't have a lot of bw to play with |
| 04:36.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:36.25 | sryan2k1 | ah, we have a little more at work, a pair of internet connections with 1mb/upload each |
| 04:36.30 | shmaltz | anybody here ever configured a tellabs echo can? |
| 04:36.58 | Lots | justinu well i have seen these other threads with other people using iax with FWD.. or do i have to download say another softphone perhaps? |
| 04:37.13 | Lots | maybe that is what i need to do eh? |
| 04:37.14 | justinu | not sure lots |
| 04:37.22 | justinu | check the docs |
| 04:37.39 | Lots | justinu you have a good suggestions for windows based softphone? |
| 04:37.50 | elvisthedj | sryan2k1: you'll have lots to choose from :) seems like more every day |
| 04:38.02 | justinu | lots: eyebeam |
| 04:38.04 | justinu | it's not free |
| 04:38.09 | justinu | unfortunately |
| 04:39.06 | shmaltz | anybody here ever configure a tellabas echo can? |
| 04:43.52 | Netgeeks | EVIL Cisco |
| 04:44.08 | Netgeeks | reporting a Mac address in decimal! |
| 04:48.53 | *** join/#asterisk techie (i=gus@antibala.com) |
| 04:50.37 | shmaltz | anybody ever configured a tellabs echo can? |
| 04:50.53 | Netgeeks | I think the silence is your answer |
| 04:51.10 | justinu | lol |
| 04:51.27 | Lots | on most softphone when interfacing them using iax to a asterisk server do you need to just put in the asterisk server, then do the registration to your provider in the asterisk iax.conf file? |
| 04:51.42 | elvisthedj | Netgeeks: i guess ResponseTimeout is gone too :/ |
| 04:51.57 | shmaltz | Netgeeks,nah you missed my earlier quoted |
| 04:52.11 | Netgeeks | I was thinking of something snide to say, but I've been in those shoes before.... I saw you ask 2 times before for a total of three |
| 04:52.12 | elvisthedj | i'm gonna take a stab and try Set(Response(timeout)=30) |
| 04:52.41 | Netgeeks | but I understand being in those shoes, so alas, I'm sorry, wish I could help, but I've never seen one, less actually configured |
| 04:53.22 | justinu | haha |
| 04:53.30 | Netgeeks | back to playing with my wonderful GUI |
| 04:53.32 | justinu | i've seen em |
| 04:53.39 | justinu | i might even have a manual for one, somewhere |
| 04:54.35 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.120) |
| 04:55.11 | justinu | that's about it |
| 04:55.26 | shmaltz | anybody here ever configured a tellabs echo can? |
| 04:55.38 | Netgeeks | ! |
| 04:56.41 | Netgeeks | are you checking in other places besides here for help, shmaltz? |
| 04:56.55 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 04:57.22 | *** part/#asterisk techie (i=gus@antibala.com) |
| 04:57.41 | Netgeeks | while I can see the connection, asking about configuring a echo can here is kinda like going to a auto-parts shop and asking about the procedure used in calibrating a torque wrench |
| 04:58.00 | Netgeeks | you might be better served finding a more appropriate forum for your question |
| 04:58.02 | elvisthedj | wow |
| 04:58.05 | elvisthedj | Jan 16 21:56:59 WARNING[11364]: pbx.c:1098 pbx_retrieve_variable: ${DATETIME} is deprecated. Please use ${STRFTIME(${EPOCH},,%m%d%Y-%H:%M:%S)} instead |
| 04:58.08 | justinu | anyone here ever configure a tellabs echo can? |
| 04:58.14 | elvisthedj | that's simpler :) |
| 04:58.17 | elvisthedj | (not) |
| 04:58.24 | justinu | lol |
| 04:58.51 | Netgeeks | No, but I slept at a holiday in express last night! how hard can it be? Let me get my easy button! |
| 04:58.57 | Netgeeks | inn |
| 04:59.12 | Netgeeks | I hate it when I ruin a punch line with bad spelling |
| 04:59.34 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 05:02.04 | shmaltz | Netgeeks of course I am |
| 05:02.27 | Netgeeks | okay, just double checking |
| 05:03.00 | Netgeeks | can't always take intelligence for granted, especially on IRC |
| 05:08.57 | samueltc | o |
| 05:11.30 | Luke-Jr | Warning to anyone planning to buy Linksys stuff: they don't honour warranties |
| 05:12.29 | joe | they have for me in the past |
| 05:14.42 | BoRiS | Anyone know how to put a default configuration in a sipura (2100) firmware image and if they try to factory reset the device, the default configuration takes presidence? |
| 05:15.57 | tengulre11 | HI,all! |
| 05:16.36 | tengulre11 | anybody know what kind of OEM X100P 's chipset?? |
| 05:18.48 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 05:18.51 | *** join/#asterisk remmo (n=rem@smack.isp.net.au) |
| 05:19.31 | *** join/#asterisk DannyF (n=dannyf@c-c982e455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 05:20.47 | dily | quit |
| 05:25.36 | *** join/#asterisk RazaMetaL (n=razameta@pc.gsalas.manta.telconet.net) |
| 05:25.38 | RazaMetaL | hi |
| 05:25.52 | RazaMetaL | good evening |
| 05:26.04 | RazaMetaL | I have a small question :) |
| 05:26.26 | RazaMetaL | i need to strip de 011 for international and replace ir for 00 |
| 05:26.40 | RazaMetaL | it's is possible ? and how ? please --- :S |
| 05:27.27 | Netgeeks | exten => _011X.,1,Dial(Blah/blah/00${EXTEN:3}) |
| 05:27.29 | ManxPower | RazaMetaL, You mean like 00${EXTEN:3} |
| 05:28.49 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:30.14 | Netgeeks | my god www.teliax.com is a busy front page |
| 05:31.04 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 05:31.45 | *** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net) |
| 05:31.50 | RazaMetaL | ManxPower, nop.. I living at south america and have one account with one US provider |
| 05:32.12 | RazaMetaL | at USA they call to international places with the 011 |
| 05:32.21 | RazaMetaL | at south america we are using 00 |
| 05:32.33 | ManxPower | so 011${EXTEN:2} then |
| 05:32.45 | ManxPower | most of the world uses 00 |
| 05:33.02 | RazaMetaL | I want to dial 001NXXNXXXXXX for USA calls |
| 05:33.15 | RazaMetaL | 0059399855138 to Ecuador :) |
| 05:33.28 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 05:35.41 | RazaMetaL | one last question .. I'm using amportal |
| 05:36.07 | RazaMetaL | how can I insert 011${EXTEN:2} with AMP _ |
| 05:36.08 | RazaMetaL | ? |
| 05:37.29 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 05:38.36 | *** join/#asterisk _Soul_ (n=Soul@87-196-39-131.net.novis.pt) |
| 05:39.07 | RazaMetaL | l33t.. the solution is here : http://sourceforge.net/forum/message.php?msg_id=3160660 |
| 05:39.12 | RazaMetaL | thankz :) |
| 05:39.30 | *** join/#asterisk c3r3al-kill3r (n=Nbm@87-196-8-26.net.novis.pt) |
| 05:41.05 | Trazzz | i have x-lite sip client and * box on my local network and a user on his own network and we both use nat. I need him to connect to my * box so we can do calls back and forth, check voicemail.. etc we can connect but can't hear anything.. no voice |
| 05:41.07 | *** join/#asterisk scolsuckz (n=scolsuck@202.58.252.15) |
| 05:56.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 05:56.46 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 05:57.28 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 05:58.30 | jaike | anyone using meetme? ive set it up and its running ok..but it doesnt seem to be recording, with the r option |
| 05:59.29 | Luke-Jr | joe: they refused to for me |
| 06:03.00 | Trazzz | i have x-lite sip client and * box on my local network and a user on his own network and we both use nat. I need him to connect to my * box so we can do calls back and forth, check voicemail.. etc we can connect but can't hear anything.. no voice |
| 06:06.35 | Luke-Jr | Trazzz: figure out STUN |
| 06:07.39 | Trazzz | Luke-Jr- do i have use STUN to make this work? |
| 06:07.44 | Trazzz | if so do you have some docs on it ? |
| 06:08.31 | *** join/#asterisk flashnet[BNC] (i=flashnet@Darkstar.AceShells.com) |
| 06:13.52 | joe | Luke-Jr: what did they refuse? Did they give you some sort of reason? |
| 06:13.54 | infinity1 | does anyone know what api is used for these polycom browser scripts? http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk |
| 06:14.01 | Luke-Jr | joe: PAP2-NA |
| 06:14.12 | infinity1 | i'm trying to figure out how to make the scripts work. |
| 06:14.27 | Luke-Jr | joe: they simply said they won't honour the warranty and expect someone else to |
| 06:16.24 | Math` | infinity1: it uses the Manager API |
| 06:16.42 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 06:17.56 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 06:20.00 | joe | Luke-Jr: whos' someone else? that's just ridiculous never heard of such a thing escalate escalate... |
| 06:20.15 | Luke-Jr | joe: the company I bought it from |
| 06:20.32 | joe | is it used? |
| 06:20.39 | joe | was it used, rather |
| 06:20.49 | Luke-Jr | before I bought it, no |
| 06:21.37 | xtrvd | I have a general setup question, does anyone mind if I ask it here? |
| 06:21.43 | joe | I just can't believe that. If you escalate and they still say that I'll be very surprised and disgusted |
| 06:23.25 | xtrvd | Alright, I'll ask in hopes that I may receive. |
| 06:23.56 | xtrvd | At the moment I am using an asterisk box with no hardware cards other than an ethernet port to connect to all my phones and to my voip provider to get service for outbound and incoming calls. I have now made the decision to go analog for incoming lines due to the quality of my provider; what hardware device will I use to provide 4 analog inputs from the telco (Wildcard TDM400P with FXO's?) and how much configuration is needed on the asterisk |
| 06:25.14 | Luke-Jr | joe: the supervisor backed it |
| 06:25.24 | wasim | xtrvd: you can use cards from digium or sangoma for fxo, configuration is minimal, two to five lines |
| 06:26.12 | joe | XTR: you set up the zap channels and you are set to go ... |
| 06:26.34 | xtrvd | wasim: Thank's for your reply. Those are the correct cards though, right? I've been trying to understand what kind of analog equipment I will need, since i Have only been using SIP/IAX over the internet so far for connectivity. |
| 06:26.35 | Luke-Jr | joe: they said it was "policy" |
| 06:26.46 | joe | policy not to honor warranty? |
| 06:26.52 | Luke-Jr | joe: and didn't care that the box says "One-Year Limited Warranty" without any exceptions |
| 06:26.56 | Luke-Jr | apparently |
| 06:26.59 | joe | Luke-Jr: what wasn't working? |
| 06:27.05 | Luke-Jr | joe: it was bricked |
| 06:27.14 | Luke-Jr | power-on self-test failed |
| 06:27.21 | wasim | xtrvd: yes, A200 from sangoma and TDM400P from digium |
| 06:27.30 | joe | Luke-Jr: tried calling at an other time and getting the name of the supervisor? |
| 06:27.38 | Luke-Jr | I have their name and badge # |
| 06:27.44 | joe | I'd call again |
| 06:27.49 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 06:27.51 | xtrvd | With my existing configuration currently serving 12 telephones (SIP), what difference in the config would be needed for ZAP channels? Or... is it ZAP that compares with IAX?... I'm a bit confused here. |
| 06:27.51 | Luke-Jr | of about 4 other employees too |
| 06:27.58 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-173.claranet.co.uk) |
| 06:28.23 | Luke-Jr | joe: I contacted the company I bought it from and they agreed to relay it, since Linksys will warranty it for them |
| 06:28.32 | Luke-Jr | just not an end user |
| 06:28.40 | joe | that's f'up |
| 06:28.46 | joe | but at least you'll get a replacement |
| 06:28.54 | joe | maybe that's what they mean by "limited" |
| 06:29.02 | joe | I dealt w/ them a lot in the past and never had an issue |
| 06:29.02 | Luke-Jr | "Limited" has a legal definition |
| 06:29.19 | Luke-Jr | as excluding natural disasters and such |
| 06:29.32 | Luke-Jr | maybe it's just -NA products |
| 06:29.44 | joe | I had hundreds of different devices from them and returned/exchaged a few dozen w/o an y issue but this was a while now |
| 06:29.48 | *** join/#asterisk Alex1 (n=A@188.90.233.220.exetel.com.au) |
| 06:29.59 | Alex1 | hello all |
| 06:30.11 | infinity1 | what are some cool things people have done with their polycom micro-browsers? |
| 06:30.35 | Alex1 | can someone point me in the right place i have just installed x100p now what do i do i need to config it |
| 06:31.21 | wasim | ~docs |
| 06:31.22 | jbot | well, 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 |
| 06:32.54 | SkramX | <3 Linux |
| 06:32.57 | SkramX | im going to take a gun to school, and... |
| 06:33.00 | SkramX | kill all the windows computers |
| 06:33.02 | SkramX | ;) |
| 06:33.31 | Alex1 | I am looking for a quick guide to config my card simple in 30min |
| 06:33.56 | SkramX | your card? |
| 06:35.36 | Alex1 | x100p |
| 06:36.01 | SkramX | Oh |
| 06:36.08 | SkramX | tried voip-info.org |
| 06:36.30 | SkramX | I do mostly virtual-asterisk stuff, am awaiting some actual phones from a business partner, sorry |
| 06:36.42 | SkramX | Alex1, as wasim said.. ~docs :) |
| 06:42.55 | Trazzz | what ports do we need open from the firewall to the * server for xlite clients? |
| 06:46.12 | xtrvd | 5060... ? |
| 06:46.49 | Trazzz | does anyone have x-lite that can test with me ? |
| 06:46.59 | xtrvd | I do, |
| 06:47.02 | xtrvd | Gotta boot up my laptop though |
| 06:47.05 | xtrvd | I'll do it for ya |
| 06:47.15 | Trazzz | ok |
| 06:47.26 | Trazzz | let me know when you are ready |
| 06:48.34 | xtrvd | Booting up now |
| 06:48.44 | xtrvd | 1 more minute til ready I suppose. |
| 06:49.16 | Trazz | i am extension 2000 |
| 06:51.08 | *** join/#asterisk lethol (n=mythtv@dsl-201-128-139-89.prod-infinitum.com.mx) |
| 06:51.14 | lethol | hi all |
| 06:51.44 | *** join/#asterisk Mark_Halverson (i=Mark_Hal@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 06:52.20 | lethol | can someone please clarify for me the span timing thing.. i read something on the wiki and zaptel.pdf on digium files say something else |
| 06:52.29 | Mark_Halverson | anyone here available to assist with an installation of GNUDIALER??? |
| 06:53.28 | Qwell | Mark_Halverson: #gnudialer |
| 06:54.26 | Mark_Halverson | thanks....m.b i made a mistake i tried connecting to it earlier |
| 06:56.23 | Mark_Halverson | got it thanks...Qwell...btw, I am looking for a custom script...to broadcast messages anyone know of a simpler prog then gnudialer or vicidialer that a dumby like me can install easily |
| 06:56.44 | lethol | nevermind just saw a post on zttool being broken.. |
| 06:59.58 | xtrvd | wasim: Which do you perfer / which have more online documentation on them, Sangoma A200 or Digium TDM400P? |
| 07:00.08 | xtrvd | Actually, that question can be for anybody. |
| 07:00.33 | Qwell | xtrvd: many of us are biased... |
| 07:01.57 | xtrvd | Qwell: What's your story? And why are you so biased? |
| 07:02.00 | xtrvd | Shareholder? =) |
| 07:02.44 | Qwell | success of Digium == success of Asterisk |
| 07:03.27 | Qwell | success of Sangoma == nothing good for Asterisk |
| 07:03.40 | Qwell | So...it's obvious where I spend my money |
| 07:04.09 | xtrvd | Touche, =) |
| 07:07.01 | *** join/#asterisk wellng (n=welles@222.90.92.113) |
| 07:07.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:12.31 | infinity1 | hmmm... for some reason presence isn't updating between my polycoms. |
| 07:13.13 | infinity1 | i chaneged my status to something else, and its not showing up on the other phone. |
| 07:15.07 | infinity1 | everything under show hints looks good. |
| 07:17.29 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:17.55 | *** join/#asterisk dasuberdavid (n=david@pcp01534754pcs.huntsv01.al.comcast.net) |
| 07:21.18 | littleball | does the order of extension definitions within the same context matter? |
| 07:24.06 | *** join/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 07:24.43 | *** join/#asterisk __Soul__ (n=Soul@87-196-13-38.net.novis.pt) |
| 07:36.14 | xtrvd | littleball: It does not. |
| 07:36.18 | xtrvd | Does not matter... |
| 07:40.09 | littleball | thanks |
| 07:40.29 | *** join/#asterisk EriSan (n=erisan@151.8.109.83) |
| 07:46.57 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 07:56.37 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:57.25 | *** join/#asterisk newlook_ (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 07:58.39 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:59.32 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
| 08:00.53 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 08:01.05 | Trazz | cool |
| 08:03.37 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 08:05.11 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 08:08.49 | *** join/#asterisk astzaphfc (n=astzaphf@bbx.lanaco.com) |
| 08:10.49 | *** join/#asterisk fugitivo (n=ajf@201.255.178.227) |
| 08:13.26 | *** join/#asterisk yxa (n=diablo@58.185.123.174) |
| 08:14.01 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 08:16.12 | astzaphfc | Hello everyone! Anyone having combination of HFC cards and TDMxxP on the samo machine? Having TDMxxP work, blocked HFC card.. Help, please! |
| 08:18.46 | enemy^x | is it possible to transfer a call from asterisk containing the original caller id? |
| 08:18.59 | h3x | enemy^x: it does that anyway. |
| 08:19.52 | enemy^x | h3x: in this situation: caller call`s in to a sip phone from pstn, sip phone redirects the call to another pstn phone, then the caller id is not kept. |
| 08:20.14 | h3x | what is your pstn connection |
| 08:20.32 | *** join/#asterisk apardo (n=apardo@87.218.44.164) |
| 08:22.09 | benjk | enem^y, it's called connected party id |
| 08:22.31 | benjk | somebody is working on a patch to do this |
| 08:22.50 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 08:26.44 | enemy^x | its a pri connection |
| 08:27.06 | enemy^x | Im starting to wonder if the telco strips of other caller ids originating on our port against them? |
| 08:28.03 | infinity1 | hmmm ..where can i get ringtones? |
| 08:28.19 | benjk | if you pick up an incoming call on a SIP phone X and then transfer the call to another SIP phone Y, then the user of Y will see the caller ID of X, not the original caller |
| 08:28.50 | benjk | so whatever your telco does or doesn't do, it's got nothing to do with this |
| 08:31.27 | *** join/#asterisk lorinc (n=ang@caracas-0974.adsl.interware.hu) |
| 08:31.36 | h3x | thats weird it seems to work fine here usually |
| 08:32.10 | h3x | does # transfer or sip transfer matter |
| 08:37.38 | *** join/#asterisk muzzz (n=chatzill@60.48.153.162) |
| 08:38.04 | *** join/#asterisk lunk (n=lunk@66.152.8.184) |
| 08:42.33 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:43.52 | *** join/#asterisk saitech (n=admin@85.235.237.14) |
| 08:44.12 | saitech | does anyone know, where i can download some free music for my "MusicOnHold" ?? |
| 08:45.43 | {zombie} | saitech: look at the voip-info.org wiki, under the musiconhold section |
| 08:45.44 | *** join/#asterisk newlook_ (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 08:45.54 | {zombie} | there's various links to royalty free classical music |
| 08:49.43 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 08:51.27 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
| 08:52.15 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 08:52.40 | *** join/#asterisk woot_ (n=woot@201.145.5.26) |
| 08:52.59 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 08:53.11 | *** part/#asterisk memic (n=memic@chicago089.server4free.de) |
| 08:53.51 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:56.14 | *** join/#asterisk gvag11 (n=g@ipa123.1.tellas.gr) |
| 08:56.19 | gvag11 | hi all |
| 08:57.44 | gvag11 | does anybody working with Tyan servers? Is GS10 reliable for TE110P solution ? |
| 08:59.02 | *** join/#asterisk veepster (i=veepster@pool-151-196-137-173.balt.east.verizon.net) |
| 08:59.15 | veepster | ccan I send a text message to an asterisk box? from a cell phone? |
| 09:01.01 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 09:01.52 | *** join/#asterisk _cleric_ (n=dacleric@p54828631.dip0.t-ipconnect.de) |
| 09:05.28 | tuxinator_linux | veepster, not a bad question. What do you want this text message to do? |
| 09:06.12 | tuxinator_linux | gvag11, I would use ASUS or Supermicro, but I am biased, as I have not used anything else |
| 09:06.12 | veepster | sit in a mysql database |
| 09:06.14 | veepster | for use later |
| 09:06.40 | veepster | but it should basically have the ability to accept text messages from cell phones..I guess they would be using sms |
| 09:06.48 | tuxinator_linux | veepster, let's check the wiki |
| 09:06.49 | veepster | but Im not going to forward the message to anyone |
| 09:06.51 | veepster | ok |
| 09:07.29 | tuxinator_linux | veepster, here is a start http://www.voip-info.org/wiki-Asterisk+cmd+Sms |
| 09:07.32 | veepster | will this need a gsm modem etc |
| 09:07.40 | veepster | thanks |
| 09:09.11 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:09.25 | *** join/#asterisk welles (n=welles@222.90.92.113) |
| 09:09.58 | h3x | heh |
| 09:10.19 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:10.31 | veepster | doesnt talk about hardware requirements.. |
| 09:10.42 | veepster | maybe there arent any |
| 09:11.08 | *** join/#asterisk pengyong (n=lala@222.188.137.148) |
| 09:11.22 | gvag11 | tuxinator : you have a bad experience with tyan ... |
| 09:11.24 | gvag11 | ? |
| 09:11.56 | veepster | I mean, the cellphone network might give up if it doesnt think the asterisk box is a cellphone, no? |
| 09:12.22 | tuxinator_linux | gvag11, I don't have any experience with tyan |
| 09:12.47 | tuxinator_linux | veepster, I'm not sure how SMS works, still learning myself |
| 09:18.36 | *** join/#asterisk CleanerX (n=nix@vpnwww01.rz.uni-karlsruhe.de) |
| 09:18.54 | veepster | thanks for the link. very helpful |
| 09:22.02 | xtrvd | Could somebody tell me what the Echo Cancellation is onboard the TDM2400E cards ? |
| 09:22.11 | xtrvd | Or even what the effects of echo are on analog ? |
| 09:24.46 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 09:25.17 | *** join/#asterisk CleanerX (n=nix@vpnwww01.rz.uni-karlsruhe.de) |
| 09:27.23 | *** join/#asterisk dc (n=nvcity@80.251.50.2) |
| 09:27.33 | *** join/#asterisk wellng (n=welles@222.90.92.113) |
| 09:28.55 | *** part/#asterisk dc (n=nvcity@80.251.50.2) |
| 09:31.50 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:34.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:37.13 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 09:38.05 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 09:40.53 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 09:41.48 | Trazzz | i am having dtmf issues.. it wont read my dtmf.. from eyebeam app.. any ideas ? |
| 09:42.53 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:45.01 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:47.25 | *** join/#asterisk goh (n=goh@60.49.6.190) |
| 09:47.52 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 09:48.32 | nfi|ermes | hi all |
| 09:49.34 | nfi|ermes | i have a sip adapter between asterisk and my normal phone (BCA) |
| 09:49.37 | jdv79 | Trazzz, has it ever worked? |
| 09:49.41 | Trazzz | nope |
| 09:49.54 | jdv79 | timing |
| 09:50.10 | Trazzz | hmm. how can i fix it? |
| 09:50.27 | nfi|ermes | in the configuration of the adapter i should configure if Flash will be sent as a DTMF <--- what do you suggest me ? |
| 09:51.02 | jdv79 | do you have zaptel hardware? i'm guessing no. |
| 09:51.30 | Trazzz | jdv79, i am sip with eyebeam softphones |
| 09:51.54 | jdv79 | setup ztdummy |
| 09:52.17 | *** join/#asterisk skamp|tee (i=skambar@p54842D13.dip0.t-ipconnect.de) |
| 09:52.23 | nfi|ermes | jdv79, if you were talking to me: i have hfc card with zaphfc driver |
| 09:52.25 | Trazzz | ? |
| 09:52.42 | *** join/#asterisk FastJack (n=fastjack@reverse-82-141-61-48.dialin.kamp-dsl.de) |
| 09:53.09 | *** part/#asterisk FastJack (n=fastjack@reverse-82-141-61-48.dialin.kamp-dsl.de) |
| 09:53.54 | jdv79 | nfi|ermes, i was talking to Trazzz. i don't know your problem. |
| 09:54.43 | Trazzz | i think ztdummy was compiled |
| 09:54.45 | Trazzz | how do i use it? |
| 09:54.47 | nfi|ermes | k |
| 09:55.47 | *** join/#asterisk cyber (n=kani@220.247.244.69) |
| 09:56.36 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 09:57.49 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:01.22 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
| 10:02.17 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:02.19 | Trazzz | anyways. any good examples on ivr ? |
| 10:05.39 | cyber | Trazzz what IVr do u need |
| 10:05.45 | cyber | u get one with asterisk |
| 10:06.09 | Trazzz | i want the system to say thank you for calling blah.com, for support hit 1 for sales hit 2, etc |
| 10:06.36 | cyber | well u do not need any thing |
| 10:06.43 | cyber | it comes with asterisk it self |
| 10:06.57 | cyber | it's the best one around |
| 10:07.06 | *** join/#asterisk alphaqu1 (n=alphaque@218.111.24.41) |
| 10:07.19 | Trazzz | wow... where do i change them or test them? |
| 10:07.34 | cyber | have u installed asterisk ? |
| 10:07.42 | Trazzz | yep its up and running |
| 10:07.50 | Trazzz | i want mine to operate just like the extesion 500 does |
| 10:08.04 | cyber | have u registered a sip phone to it |
| 10:08.13 | Trazzz | other than have the lady say my name |
| 10:08.22 | Trazzz | yes i have phone registered to it |
| 10:08.29 | cyber | can u dial 1000 |
| 10:08.33 | cyber | on the phone |
| 10:08.42 | Trazzz | yes |
| 10:08.49 | cyber | what happen |
| 10:08.53 | cyber | do u get the IVR |
| 10:09.03 | Trazzz | yes she is still talking |
| 10:09.07 | *** join/#asterisk wilymage (n=wily@funkmunch.net) |
| 10:09.15 | cyber | it says press 1 and 2 |
| 10:09.23 | cyber | u need some thing like this right |
| 10:09.29 | Trazzz | yes exactly |
| 10:09.34 | *** part/#asterisk alphaqu1 (n=alphaque@218.111.24.41) |
| 10:09.42 | cyber | check the extentions.conf file |
| 10:09.51 | Trazzz | also can i do my whole system in her voice? |
| 10:09.53 | cyber | u have to configure ur IVR there |
| 10:10.04 | *** join/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 10:10.07 | Trazzz | so i configure the ivr in that menu or via files? |
| 10:10.07 | cyber | yes u can record voice files and upload |
| 10:10.28 | cyber | r u new to asterisk |
| 10:10.29 | Trazzz | any way to have her voice ? |
| 10:10.33 | Trazzz | yes i am new to asterisk |
| 10:10.52 | Trazzz | i am playing with extensions.conf |
| 10:10.55 | Trazzz | and sip.conf |
| 10:10.59 | wilymage | hey guys, having troubles with music on hold. think it's a timer issue with asterisk (using snom 360s). i've read the forums, had techs try everything, i'm on the mailing lists. tried writing a script that pipes to aplay instead of mpg123, still no dice. festival works fine. moh is still distorted, untimed, clicking, etc. need to compile with zaptel support? help! |
| 10:11.27 | cyber | i remeber me being in ur stage and how hard it was to get help, i will help u |
| 10:11.34 | cyber | open extentions.conf |
| 10:11.44 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:11.54 | Trazzz | ok much appreciated |
| 10:11.55 | cyber | in /etc/asterisk |
| 10:12.01 | Trazzz | got it open |
| 10:12.53 | cyber | 1 min |
| 10:13.33 | *** join/#asterisk nicksin (n=nicl@209.212.104.185) |
| 10:13.39 | nicksin | Good morning all |
| 10:14.12 | cyber | ok lets start |
| 10:14.16 | Trazzz | ok |
| 10:14.25 | cyber | first let's create a new context for u |
| 10:14.43 | Trazzz | ok |
| 10:14.48 | cyber | add the following to the end of the extentions.conf |
| 10:14.57 | cyber | [myivr] |
| 10:15.09 | Trazzz | ok got it |
| 10:15.36 | cyber | now open sip.conf and under ur sip user |
| 10:15.43 | cyber | add the following |
| 10:15.47 | cyber | context=myivr |
| 10:16.05 | *** join/#asterisk _Soul_ (n=Soul@87-196-13-38.net.novis.pt) |
| 10:16.19 | Trazzz | ok |
| 10:16.47 | cyber | ok now go back to extentions.conf |
| 10:17.20 | Trazzz | ok |
| 10:17.25 | cyber | how do u like ur IV |
| 10:17.30 | cyber | IVR |
| 10:18.32 | Trazzz | ? |
| 10:18.44 | Trazzz | i need to reload right? |
| 10:18.46 | cyber | what do u want when some one press 1 |
| 10:18.48 | cyber | and 2 |
| 10:18.56 | cyber | we r not done yet |
| 10:18.57 | Trazzz | yes press 1 for sales, press 2 for support |
| 10:18.58 | goh | may I know 2 ISDN connection need to use with type of digium card? |
| 10:19.21 | cyber | sales and support and sip phones ? |
| 10:19.25 | Trazzz | yes |
| 10:19.31 | Trazzz | and 0 for operator |
| 10:19.35 | *** join/#asterisk Alex1 (n=A@188.90.233.220.exetel.com.au) |
| 10:19.43 | Alex1 | hello all |
| 10:20.02 | cyber | ok give me ur email address i will send u some code paste it under [myivr] |
| 10:21.10 | Alex1 | I'm haveing this problem i just installed astlinux and i am making changes i change the file system to reade/write i make my changes save them reboot and its default again !!! HELP .. |
| 10:21.49 | dudes | check /etc/fstab |
| 10:22.10 | Trazz | did you get my pm cyber ? |
| 10:22.12 | *** join/#asterisk fulgas (n=fulgas@209.8.233.244) |
| 10:22.51 | Alex1 | also if i use bash: when i reboot there is no history :( how can i fix this Astlinux is installed to my hdd |
| 10:23.23 | dudes | Perhaps make the filesystem rw |
| 10:23.43 | Alex1 | i do that in the web admin |
| 10:23.53 | Alex1 | whats going on |
| 10:24.11 | Alex1 | how to i change it for good in the command line |
| 10:25.39 | dudes | man fstab at the command line maybe? You could also try google or the wiki. |
| 10:26.37 | Trazz | great page for anyone with SIP and NAT |
| 10:26.37 | Trazz | http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 10:26.59 | Trazz | and this one |
| 10:27.00 | Trazz | http://support.counterpath.com/viewtopic.php?t=5236&highlight=voice&sid=e55d0f192847e28dfe942da473410819 |
| 10:28.30 | *** join/#asterisk vladuk (n=vk@ip.82.144.198.127.stat-2.volia.net) |
| 10:28.41 | vladuk | HI. All |
| 10:28.53 | Trazz | Cyber, you still here? |
| 10:29.38 | vladuk | Have a problem receiving DTMF from cell phones when call comes via land line. |
| 10:32.24 | vladuk | is here anyone who could shed a light on that issue ? |
| 10:33.35 | Trazz | ? |
| 10:33.56 | cyber | trazz and wilymage |
| 10:33.59 | cyber | ur mails sent |
| 10:34.07 | cyber | check mail and enjoj |
| 10:34.10 | cyber | enjoy |
| 10:34.47 | Trazz | ah ok.. thanks le tme check it |
| 10:35.19 | *** join/#asterisk heroine (n=heroine@lit75-2-82-225-244-34.fbx.proxad.net) |
| 10:35.32 | Alex1 | how do i change astlinux to reade/write |
| 10:36.09 | *** part/#asterisk heroine (n=heroine@lit75-2-82-225-244-34.fbx.proxad.net) |
| 10:36.37 | *** join/#asterisk lahaine (n=lahaine@63.66.119-80.rev.gaoland.net) |
| 10:36.43 | lahaine | hi |
| 10:36.49 | cyber | asterisk even supports having one than one support sip users and tring them one by one untill a user answers |
| 10:36.53 | Trazz | gonna try now.. thanks very very much :) |
| 10:37.04 | cyber | np |
| 10:37.25 | cyber | u have record ur own gsm ivr promts and upload |
| 10:37.39 | cyber | and change the name in the extentions |
| 10:37.41 | Alex1 | how do i change astlinux to reade/write |
| 10:38.25 | vladuk | mount -o remount /partition rw |
| 10:40.32 | Trazz | cyber, is that option 3 for operator instead of hitting 0 ? |
| 10:40.55 | cyber | yup |
| 10:40.58 | cyber | have it 0 |
| 10:41.04 | cyber | replace 3 with 0 |
| 10:41.11 | cyber | then u r done |
| 10:42.29 | Trazz | it says that is option is not available when i hit 1 or 0 even after reload |
| 10:42.54 | cyber | have u registered the sip user ? |
| 10:42.59 | cyber | for support |
| 10:43.02 | Alex1 | hello you know in the webmin that astlinux provides i change the file system to reade/write i make and commit changes reboot and nothings there !!!!!!! |
| 10:43.18 | Trazz | yes |
| 10:43.26 | Trazz | i had them relogin |
| 10:43.40 | Trazz | after changes |
| 10:44.20 | cyber | does it goes to vice mail ? |
| 10:44.27 | cyber | voice mail ? |
| 10:44.41 | Trazz | no |
| 10:44.46 | Trazz | that is option is invalid |
| 10:44.48 | Trazz | weird |
| 10:45.08 | Alex1 | can someone please help me make my hdd reade/write please ?? |
| 10:45.42 | cyber | do u get the instructions |
| 10:45.56 | Alex1 | mount -o remount /partition rw |
| 10:46.00 | vladuk | forget about webmin, U don't need it |
| 10:46.02 | Alex1 | the rw part dose not work |
| 10:46.07 | cyber | does it work for 2 |
| 10:46.21 | MrChimpy | asterisk mailing listst still down? |
| 10:46.33 | Alex1 | cyber |
| 10:46.35 | Alex1 | ? |
| 10:46.38 | Trazz | yep option 2 works |
| 10:46.57 | Alex1 | i lost now |
| 10:47.20 | cyber | call comes to sales ? |
| 10:47.39 | Trazz | let me test it again |
| 10:47.44 | Alex1 | this must be realy simple please help |
| 10:48.42 | Trazz | it plays the what is * |
| 10:48.53 | Trazz | * is an opensource |
| 10:48.59 | Trazz | it doesnt hit the sales extension |
| 10:49.04 | Trazz | just plays the greeting |
| 10:49.13 | Trazz | talks about hitting 2 or 500 |
| 10:49.22 | cyber | ahh |
| 10:49.32 | cyber | did u dial 2000 |
| 10:49.38 | tuxinator_linux | demo on? |
| 10:49.40 | Trazz | yes |
| 10:49.58 | Trazz | i dialed 2010 which is what i changed it to |
| 10:50.00 | Trazz | it answers |
| 10:50.12 | Alex1 | tuxinator_linux: you talking to me |
| 10:50.20 | cyber | wait a min |
| 10:50.39 | tuxinator_linux | piping in, not really knowing what I am talking about. |
| 10:50.42 | Trazz | ok |
| 10:50.48 | vladuk | Alex1, just execute from root console: mount and check options for necessary partition |
| 10:52.03 | cyber | u r getting the default asterisk voice promts |
| 10:52.09 | Trazz | yes |
| 10:52.11 | cyber | that's why u get 500 and 600 |
| 10:52.18 | cyber | u have to record ur own and replace |
| 10:52.29 | cyber | just ignore them and press 1 |
| 10:52.39 | Trazz | yep, but it should transfer to my extensionos.. |
| 10:52.45 | Trazz | when i hit one its not available.. |
| 10:53.12 | cyber | do u get a voice message |
| 10:53.23 | Trazz | yes the lady says it not available |
| 10:54.01 | Trazz | Playing 'demo-congrats' (language 'en') |
| 10:54.01 | Trazz | <PROTECTED> |
| 10:54.01 | Trazz | <PROTECTED> |
| 10:54.01 | Trazz | <PROTECTED> |
| 10:54.01 | Trazz | <PROTECTED> |
| 10:54.04 | cyber | trazz can u give me 10 mins |
| 10:54.08 | Trazz | sure thing |
| 10:54.20 | thazza | Please pick an option.. 1 for sales, 2 for support. or headbutt the wall for immedite transfer. (Headbutt) Your call has been queued to the lost souls deparment. Please hold... |
| 10:54.21 | cyber | i'll fix this for u in 10 mins |
| 10:54.26 | Trazz | sure thing.. thanks |
| 10:56.29 | vladuk | is there anyone knows anything about DTMF processing ? |
| 10:57.16 | vladuk | I have such schema: asterisk <- sipura3000 <- Landline |
| 10:57.58 | thazza | vladuk: Thats the setup i have working at home. :D |
| 10:58.04 | vladuk | who is dealing with DTMF sipura3000 or Asterisk ? |
| 10:58.05 | nfi|ermes | Trazz, where re u from ? |
| 10:58.57 | thazza | vladuk: Yet mine is more like this.. Asterisk <-> Sipura3000 <-> Laneline. |
| 10:59.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:00.50 | puzzled | morning all |
| 11:00.59 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 11:02.12 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:02.15 | thazza | morning puzzled |
| 11:03.58 | goh | who can teach me how to add iax fwd.pulver as trunks in AMP |
| 11:04.20 | goh | I follow asterisk@home hand books in does not work |
| 11:06.15 | thazza | ~amp |
| 11:06.17 | jbot | i guess amp is NOT supported here! people using it should join #amportal |
| 11:06.33 | wilymage | amp is ugly |
| 11:07.17 | *** join/#asterisk tameristx (n=arkadasc@85.97.21.34) |
| 11:07.30 | [av]bani | what's a better webinterface for asterisk? |
| 11:07.39 | lahaine | i'm trying to get realtime working throught odbc to load sip peers but i cannot get it working thought i've been trying a couple of configurations. |
| 11:08.43 | thazza | [av]bani: vi |
| 11:09.05 | lahaine | it seems that it connect suffessfully to the mysql database throught odbc but no queries are issued |
| 11:10.12 | jaike | thazza: hehe |
| 11:10.56 | thazza | lahaine: Have you put the correct info in the res_odbc.conf file in /etc/asterisk ? |
| 11:11.04 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 11:12.20 | lahaine | thazza, i think so |
| 11:13.20 | fenlander | hi, does anyone here have any experience with SendURL()? Does it work with SIP channels? |
| 11:13.39 | h3x | fenlander: no |
| 11:13.43 | h3x | sip dosent have that concept. |
| 11:13.44 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 11:13.50 | h3x | yet |
| 11:14.00 | fenlander | h3x: okay - thanks - I thought that was the case |
| 11:14.19 | lahaine | thazza, if you can tell me if that conf seems to appear correct : http://pastebin.com/509668 |
| 11:14.36 | h3x | and the only iax phone that actually works with it is diax phone |
| 11:14.46 | h3x | and gnophone which is a piece of crap |
| 11:14.52 | thazza | i must admit, that sendurl() is a very interesting feature i will be looking at soon.. :D Heaps of ideas for that one. |
| 11:16.31 | Trazz | what does everyone use for extensions normally on * ? |
| 11:16.43 | NewSole | http://www.killsometime.com/animations/Files-1952196/One-Handed.swf |
| 11:17.58 | thazza | lahaine: Looks good to me.. :D |
| 11:18.05 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF68FB.dip.t-dialin.net) |
| 11:18.14 | lahaine | damn ... |
| 11:18.34 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:19.03 | lahaine | configuration from the database should be loaded at asterisk startup ? or when data is needed by a user action like passing a call ? |
| 11:19.55 | thazza | NewSole: Does it do anything other than load? |
| 11:20.06 | lahaine | from what i understood, if it's the static realtime configuration, with sip.conf, it should be loaded on startup whereas using a family name makes the loading dynamic when needed, is that right ? |
| 11:21.04 | Skymarshal | Hi, I don't understand the following problem: |
| 11:21.05 | Skymarshal | I want to make a call to SIP:007@company.com from my SNOM 360 which is connected to my Asterisk. Does the SNOM connects direct to the company.com Asterisk server or do I have to impliment an extensions.conf rule for that? |
| 11:21.23 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 11:21.45 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:23.02 | *** join/#asterisk Ikarus__ (n=ikarus@c109100.upc-c.chello.nl) |
| 11:26.22 | *** join/#asterisk RoyK (n=roy@213.160.242.42) |
| 11:26.22 | *** join/#asterisk cyber (n=kani@220.247.244.36) |
| 11:26.40 | cyber | hi i am back |
| 11:30.24 | *** join/#asterisk cyber (n=kani@220.247.244.36) |
| 11:31.01 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:31.12 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
| 11:34.05 | *** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee) |
| 11:35.29 | [av]bani | thazza: vi is a webinterface? |
| 11:35.41 | Skymarshal | Anyone? Please. |
| 11:35.42 | Skymarshal | I have a running Asterisk PBX. But until now I'ved used it just for "normal" telephone calls. Now I want to add SIP and AIX connections. But somehow I don't know how to do this. Does the SIP-Phone creates a connection to the SIP-Server of the SIP-Phone-Adress or does is connects to my Asterisk and it has to bypass it? |
| 11:36.27 | cyber | thazza u ready to fix ur IVR |
| 11:36.37 | lahaine | you have to setup some sip users in your sip.conf file |
| 11:36.52 | thazza | cyber: Trazz left. |
| 11:37.02 | cyber | ohh ok |
| 11:37.20 | cyber | :) |
| 11:37.31 | thazza | [av]bani: Yep ;-) |
| 11:38.01 | Skymarshal | lahaine: why? I don't understand your point. |
| 11:38.39 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 11:38.54 | lahaine | Skymarshal: hmm .. i'm not sure to understand well what you want to do in fact |
| 11:39.51 | cyber | hey any of u guys had luck registering a sip phone which is behind proxy |
| 11:39.55 | jaike | skymarshall: does your snom register properly to your asterisk server? as a sip client |
| 11:40.16 | jaike | cyber: big problem..depends on the proxy |
| 11:40.38 | Skymarshal | lahaine: right know i have a |
| 11:40.40 | Skymarshal | exten => _0[1-9].,1,Dial(SIP/${EXTEN}@my-account,120) |
| 11:40.42 | Skymarshal | line in my extensions.conf and that is fine for all my normal calls. But who can rute a SIP-Call. I know it must be a simple problem. But I am stuck. |
| 11:41.42 | cyber | yup i use winproxy only thing is there's no place to enter a proxy address(not the SIP proxy) in the ip phone |
| 11:43.54 | *** join/#asterisk benjk_ (n=benjamin@AI-202-179-202-225.tik.access-internet.ne.jp) |
| 11:44.11 | lahaine | Skymarshal, cannot help you :/ |
| 11:44.30 | Skymarshal | lahaine: thanks anyway. |
| 11:45.11 | cyber | skymarshal what the problem |
| 11:46.57 | Skymarshal | cyber: until know I've used my Asterisk for "normal" call through a SIP-Provider Gateway only. But I'd like to connect to other SIP-Phones direkt. So e.g. I want to call SIP:021123704140@company.com and have no idea where and how to put this into my extensions.conf. |
| 11:49.21 | cyber | well that's easy |
| 11:49.24 | cyber | hold |
| 11:51.24 | cyber | exten => 2000,1,Dial(SIP/user@domain.com) |
| 11:53.30 | Modcuts | Is there a dial rule i can use to call internationally with lots of different countries instead of inputing the dial rules for each country? |
| 11:53.58 | Skymarshal | cyber: thanks! |
| 11:54.17 | cyber | i am already doing this |
| 11:54.35 | cyber | but the problem is some sip providers block ips |
| 11:54.58 | cyber | if so u have to ask them to enable ur ip |
| 11:57.28 | lahaine | i'm still stuck with the realtime configuration ... SQLExecute from the odbc module are never called, wether at sartup wether when i use a realtime load voicemail email %user% :/ |
| 11:57.37 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 11:58.13 | *** join/#asterisk xyu (n=user@195.206.49.114) |
| 11:58.17 | xyu | hello! |
| 11:58.23 | *** join/#asterisk Atomik (n=pIRCuser@195.206.49.114) |
| 11:58.27 | Atomik | 123 |
| 11:58.33 | benjk | Skymarshal: we have a HOWTO for that on http://www.astmasters.net/howtos.html |
| 11:58.50 | benjk | scroll down to where it says "SIP URIs" |
| 11:58.51 | xyu | hey bitches? |
| 11:59.01 | trixter | moo |
| 11:59.02 | Atomik | twisted[asteria] kram drumkilla denon anthm whom bot ? |
| 11:59.05 | *** join/#asterisk p9goBou_LLlyTHuK (n=iNfErNo@80.81.208.34) |
| 11:59.07 | benjk | ??? |
| 11:59.09 | xyu | trixter |
| 11:59.10 | xyu | benjk |
| 11:59.11 | xyu | trixter |
| 11:59.17 | p9goBou_LLlyTHuK | lol |
| 11:59.19 | xyu | 8:13] * Nae?an Au ia eaiaea #asterisk |
| 11:59.20 | xyu | [19:58:13] * Oiiee 'Asterisk 1.2.1 has been released! -//- http://www.asterisk.org' |
| 11:59.20 | xyu | [19:58:13] * Onoaiiaeai drumkilla on Thu Dec 08 03:04:08 |
| 11:59.20 | xyu | [19:58:16] <xyu> hello! |
| 11:59.20 | xyu | [19:58:23] * Atomik (n=pIRCuser@195.206.49.114) i?eoiaeo a #asterisk |
| 11:59.20 | xyu | [19:58:27] <Atomik> 123 |
| 11:59.22 | trixter | um hi |
| 11:59.22 | xyu | [19:58:33] <benjk> Skymarshal: we have a HOWTO for that on http://www.astmasters.net/howtos.html |
| 11:59.23 | p9goBou_LLlyTHuK | OO |
| 11:59.26 | p9goBou_LLlyTHuK | xaxaxa |
| 11:59.39 | p9goBou_LLlyTHuK | :D |
| 11:59.40 | trixter | what the hell was that? |
| 11:59.45 | Atomik | Albania 4rever |
| 11:59.51 | p9goBou_LLlyTHuK | :D |
| 11:59.54 | p9goBou_LLlyTHuK | :D |
| 11:59.57 | p9goBou_LLlyTHuK | niibaca |
| 12:00.04 | *** join/#asterisk pengyong (n=lala@222.185.197.28) |
| 12:00.07 | p9goBou_LLlyTHuK | <anthm>, <denon>, <drumkilla>, <kram>, <twisted[asteria]>, dodz |
| 12:00.08 | Atomik | <PROTECTED> |
| 12:00.12 | benjk | what's this? #kindergarden |
| 12:00.19 | p9goBou_LLlyTHuK | tebe tuda |
| 12:00.23 | p9goBou_LLlyTHuK | nada |
| 12:00.32 | p9goBou_LLlyTHuK | <benjk>, eblan |
| 12:00.33 | *** join/#asterisk xyu (n=user@195.206.49.114) |
| 12:00.36 | xyu | 123 |
| 12:00.36 | p9goBou_LLlyTHuK | <benjk>, eblan |
| 12:00.37 | Atomik | anthm denon drumkilla kram twisted[asteria] so whom bots ? |
| 12:00.37 | *** join/#asterisk RoadRunnR (n=MrRoadRu@213.187.82.17) |
| 12:00.38 | xyu | test |
| 12:00.39 | p9goBou_LLlyTHuK | :D |
| 12:00.42 | xyu | beta test |
| 12:00.46 | Atomik | =) |
| 12:00.46 | p9goBou_LLlyTHuK | hmmm |
| 12:00.50 | xyu | alpha test |
| 12:00.51 | p9goBou_LLlyTHuK | yuuu |
| 12:00.54 | xyu | it's working |
| 12:01.11 | Atomik | äà âû ÷¸ ïèäîðàñû ñàâñåì îõóåëè ñèäÿò 400 ÷åë è âñå ìîë÷àò |
| 12:01.13 | Atomik | äà âû ÷¸ ïèäîðàñû ñàâñåì îõóåëè ñèäÿò 400 ÷åë è âñå ìîë÷àò |
| 12:01.16 | VeNoM | àãà |
| 12:01.19 | xyu_like_flood | ïèäàðû |
| 12:01.22 | xyu_like_flood | åáëèâûå |
| 12:01.24 | VeNoM | óèèèèèèè |
| 12:01.28 | xyu_like_flood | õóëè çàãàñèëèñü |
| 12:01.31 | xyu_like_flood | ïîä ëàôêè |
| 12:01.34 | Atomik | <PROTECTED> |
| 12:01.37 | xyu_like_flood | õóèïëåòû |
| 12:01.44 | VeNoM | ÎÎ |
| 12:01.46 | xyu_like_flood | AndyCap ïèäîð |
| 12:01.49 | xyu_like_flood | A-Tuin õóé |
| 12:01.54 | xyu_like_flood | ApEtc øëþõà |
| 12:01.56 | VeNoM | =) |
| 12:01.59 | xyu_like_flood | arguile õóëè ðàññåëñÿ |
| 12:01.59 | Atomik | ApEtc ñóêà |
| 12:02.04 | iDunno | what the fuck. |
| 12:02.06 | xyu_like_flood | chapeaurouge èäè íàõóé |
| 12:02.09 | VeNoM | <denon>, <drumkilla>, <kram>, <twisted[asteria]>, åáëàíû |
| 12:02.10 | iDunno | please be behaving. |
| 12:02.17 | VeNoM | <iDunno>, íàõóé èäè |
| 12:02.17 | xyu_like_flood | iDunno uck my dick bitch |
| 12:02.18 | Atomik | iDunno you fuck!!! |
| 12:02.21 | xyu_like_flood | iDunno suck my dick bitch |
| 12:02.36 | VeNoM | îÎ |
| 12:02.37 | iDunno | grow the fuck up. get a life. |
| 12:02.40 | NewSole | iDunno.... it just shows their mentality.... |
| 12:02.50 | VeNoM | nea |
| 12:02.52 | xyu_like_flood | NewSole go fuck |
| 12:03.00 | xyu_like_flood | NewSole stupid noob |
| 12:03.02 | benjk | that's it final confirmation that #asterisk == #kindergarden |
| 12:03.06 | VeNoM | <NewSole>, ïðîñòî íàì äåëàòü íåõóé |
| 12:03.08 | lahaine | sad brain can't be sold to replace deffective ones |
| 12:03.10 | NewSole | they probly have not ever been laied... so they get off on computer flooding |
| 12:03.20 | xyu_like_flood | flood |
| 12:03.23 | Atomik | NewSole à òû âàùå åáàëî çàâàëè |
| 12:03.25 | xyu_like_flood | it is flood |
| 12:03.26 | xyu_like_flood | it is flood |
| 12:03.29 | xyu_like_flood | it is flood |
| 12:03.30 | *** part/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:03.30 | xyu_like_flood | it is flood |
| 12:03.44 | xyu_like_flood | it is flood |
| 12:04.00 | xyu_like_flood | it is flood |
| 12:04.00 | NewSole | thats right yanki go home... I live in canada |
| 12:04.01 | Atomik | <PROTECTED> |
| 12:04.01 | MrChimpy | anyone fancy opping me? |
| 12:04.02 | Atomik | <PROTECTED> |
| 12:04.07 | Atomik | <PROTECTED> |
| 12:04.11 | Atomik | <PROTECTED> |
| 12:04.12 | *** mode/#asterisk [+b *!*@80.81.208.34] by kram |
| 12:04.16 | *** kick/#asterisk [VeNoM!n=mark@pdpc/sponsor/digium/kram] by kram ((*@80.81.208.34) CLONEBOTS.. AUTO KICK ACTIVATED) |
| 12:04.21 | MrChimpy | thanks kram |
| 12:04.24 | anderiv | heh |
| 12:04.26 | Atomik | <PROTECTED> |
| 12:04.34 | xyu_like_flood | l] |
| 12:04.35 | xyu_like_flood | [ |
| 12:04.40 | Atomik | you all bots |
| 12:04.50 | NewSole | USSR = User Sucks Shit Realtime |
| 12:04.51 | xyu_like_flood | ][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][] |
| 12:04.52 | xyu_like_flood | ][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][] |
| 12:04.56 | xyu_like_flood | ][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][][ |
| 12:04.59 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:05.03 | Atomik | you all bots |
| 12:05.11 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 12:05.23 | MrChimpy | hmm. IRC is a special place. |
| 12:05.27 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:05.30 | MrChimpy | special as in special school. |
| 12:05.35 | trixter | ignore works well also |
| 12:05.45 | xyu_like_flood | sorry for my bad english |
| 12:05.53 | NewSole | like I said |
| 12:05.54 | NewSole | USSR = User Sucks Shit Realtime |
| 12:06.01 | NewSole | USSR = User Sucks Shit Realtime |
| 12:06.03 | Atomik | NewSole aaa |
| 12:06.03 | NewSole | USSR = User Sucks Shit Realtime |
| 12:06.07 | iDunno | MrChimpy: no, IRC is a fine place, it's these damned skript kiddies that have nothing better today because they're fucking morons that are the problem. |
| 12:06.09 | xyu_like_flood | sorry for my bad english |
| 12:06.11 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:06.11 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:06.12 | Atomik | NewSole you bot |
| 12:06.13 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:06.17 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:06.17 | iDunno | s/today/todo/ |
| 12:06.19 | xyu_like_flood | NewSole USSR fuck you in your ass hole when you be a kid |
| 12:06.26 | lahaine | Is there a way to test a realtime configuration step by step .. ? |
| 12:06.33 | xyu_like_flood | USA - Users Suck Ass |
| 12:06.36 | *** part/#asterisk chet (n=chet@71-212-36-63.tukw.qwest.net) |
| 12:06.42 | NewSole | lahaine> no |
| 12:06.43 | trixter | who is the bigger moron the one who floods or the ones who dont use ignore, built into every irc client for a decade? :P |
| 12:06.52 | benjk | kram: please kick xyu too |
| 12:06.58 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.00 | xyu_like_flood | no |
| 12:07.04 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.05 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.06 | *** join/#asterisk eivindtr (n=wingnut-@217.68.103.66) |
| 12:07.06 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.13 | trixter | lahaine: that would be a nice tool it might help fix some code problems but I dont think 'testing' is part of the asterisk model sadly |
| 12:07.13 | xyu_like_flood | USA must suck |
| 12:07.14 | xyu_like_flood | USA must suck |
| 12:07.16 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.16 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.18 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.19 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.20 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.20 | Atomik | USA MUST DIE!!!!!!!! |
| 12:07.22 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.24 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.25 | puzzled | trixter: hehe |
| 12:07.26 | xyu_like_flood | USA - Users Suck Ass |
| 12:07.30 | xyu_like_flood | USA must suck |
| 12:07.32 | lahaine | NewSole,trixter> huh :'( |
| 12:07.32 | xyu_like_flood | USA must suck |
| 12:07.34 | xyu_like_flood | USA must suck |
| 12:07.34 | *** part/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:07.35 | Atomik | FUCKING ENGLISH!!! |
| 12:07.36 | xyu_like_flood | USA must suck |
| 12:07.39 | Atomik | FUCKING ENGLISH!!! |
| 12:07.40 | xyu_like_flood | trixter |
| 12:07.40 | Atomik | FUCKING ENGLISH!!! |
| 12:07.42 | xyu_like_flood | puzzled |
| 12:07.46 | xyu_like_flood | trixter |
| 12:07.47 | Atomik | FUCKING ENGLISH!!! |
| 12:07.48 | xyu_like_flood | xyu |
| 12:07.52 | MrChimpy | what's the ignore command in x-chat? /ignore seems to work by host |
| 12:07.54 | xyu_like_flood | xyu |
| 12:07.58 | xyu_like_flood | xyu BaM B poT |
| 12:08.04 | MrChimpy | i hardly ever use IRC. use telnet based talkers |
| 12:08.04 | xyu_like_flood | xyu BaM B poT |
| 12:08.04 | lahaine | right click on the nick MrChimpy |
| 12:08.05 | trixter | right click the username, select ignore, |
| 12:08.06 | xyu_like_flood | xyu BaM B poT |
| 12:08.08 | xyu_like_flood | xyu BaM B poT |
| 12:08.10 | xyu_like_flood | xyu BaM B poT |
| 12:08.11 | Atomik | MrChimpy ÎÁÑÎÑÈ ÌÎÞ ÆÎÏÓ !!!!!!!!!!! |
| 12:08.12 | xyu_like_flood | xyu BaM B poT |
| 12:08.14 | xyu_like_flood | xyu BaM B poT |
| 12:08.16 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.18 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.20 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.20 | Atomik | MrChimpy ÎÁÑÎÑÈ ÌÎÞ ÆÎÏÓ !!!!!!!!!!! |
| 12:08.22 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.22 | Atomik | MrChimpy ÎÁÑÎÑÈ ÌÎÞ ÆÎÏÓ !!!!!!!!!!! |
| 12:08.24 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.25 | Atomik | MrChimpy ÎÁÑÎÑÈ ÌÎÞ ÆÎÏÓ !!!!!!!!!!! |
| 12:08.26 | xyu_like_flood | MrChimpy go fuck lame |
| 12:08.27 | Atomik | MrChimpy ÎÁÑÎÑÈ ÌÎÞ ÆÎÏÓ !!!!!!!!!!! |
| 12:08.28 | xyu_like_flood | trixter lame lame lame |
| 12:08.30 | MrChimpy | ah! peace! |
| 12:08.30 | xyu_like_flood | trixter lame lame lame |
| 12:08.32 | xyu_like_flood | trixter lame lame lame |
| 12:08.34 | puzzled | indeed |
| 12:08.34 | xyu_like_flood | trixter lame lame lame |
| 12:08.35 | Atomik | MrChimpy íó òû ÷óðêà äàæå ðóñêèé íå ïîíèìàåø |
| 12:08.36 | xyu_like_flood | trixter lame lame lame |
| 12:08.36 | trixter | heh |
| 12:08.37 | MrChimpy | I wasn't expecting guis :) |
| 12:08.38 | xyu_like_flood | trixter lame lame lame |
| 12:09.20 | xyu_like_flood | buy buy stupid girls |
| 12:09.22 | NewSole | lahaine> what is the problem |
| 12:09.23 | *** part/#asterisk xyu_like_flood (n=user@195.206.49.114) |
| 12:10.22 | Atomik | NewSole äà èäè òû íàõóé ñóêà !!! |
| 12:10.48 | lahaine | the problem is that nothing is loaded from the database tables . i've tried to load sip buddies, voicemail, extension dialplan .. and i cannot get anything loaded .. i assume i've forgot some silly option somewhere but can't figure out wich one |
| 12:11.04 | NewSole | no I wanted to use the "Wet Tampon" one |
| 12:12.01 | lahaine | the only thing i'm quite sure about is that the odbc configuration is ok as i get conection from asterisk in mysql log |
| 12:12.54 | NewSole | problem is there is a lot of files to check for realtime |
| 12:13.35 | *** part/#asterisk SadamHysein (n=pIRCuser@195.206.49.114) |
| 12:13.41 | lahaine | from the asterisk loading output it seems to be weel congfigured as i get those messages : == Binding sippeers to odbc/ myastdb/ sip_buddies |
| 12:14.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:14.19 | lahaine | my res_odbc.conf seems to be ok |
| 12:14.48 | NewSole | extconfig.conf |
| 12:15.04 | NewSole | and odbc.ini and odbcinst.ini |
| 12:16.17 | trixter | I think that part of the problem with making good test tools is the way that asterisk is done. Some of the stuff would seem better suited for a library, which if done could mean that simple test tools could make the calls to that lib and test functionality better.. of course this requires a bunch of rewriting to make that a reality, but still.. The way it is now the daemon, cli, etc are all the same program (generally refered to as a bad design) an |
| 12:16.17 | trixter | d the way its all linked together it makes it harder to test components ... a library with a small glue program to make the libraries do something would greatly improve testing (assumnig proper test cases were made) and make it easier for people to create patches |
| 12:16.35 | trixter | but what do I know, I have only been coding for 27 years |
| 12:16.47 | *** join/#asterisk Jedirl (n=hhgds4@213.162.200.226) |
| 12:17.06 | lahaine | files are there : http://pastebin.com/509668 |
| 12:17.34 | *** join/#asterisk CuTe^G (n=cuteg@202.63.195.128) |
| 12:17.52 | CuTe^G | hello brother i installed Asterisk@Home 1.5 |
| 12:18.09 | CuTe^G | from which command i will set the IP ? |
| 12:18.17 | NewSole | lahaine> starters... remove the extra spaces.... |
| 12:18.34 | trixter | ifconfig? |
| 12:18.41 | NewSole | sipusers => odbc, myastdb, sip_buddies |
| 12:18.45 | NewSole | sould be |
| 12:18.51 | NewSole | sipusers => odbc,myastdb,sip_buddies |
| 12:19.10 | lahaine | hmm ok .. |
| 12:19.15 | NewSole | dummies did not program adaptive spacing |
| 12:19.23 | CuTe^G | i installed Asterisk@Home 1.5 ... how i can set the IP in server to open at client computer |
| 12:19.42 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:19.54 | lahaine | damn, thanks ! |
| 12:20.08 | NewSole | did that work |
| 12:20.12 | lahaine | i've enough of those fucking silly parser ... ! |
| 12:20.19 | lahaine | yes, it works better :) |
| 12:20.51 | *** join/#asterisk tm (i=tm@mebes.info) |
| 12:20.55 | tm | hello. |
| 12:21.01 | CuTe^G | i installed Asterisk@Home 1.5 ... how i can set the IP in server to open at client computer |
| 12:21.02 | NewSole | u have to remember... asterisk works well but its a hack job |
| 12:21.12 | tm | kann hier wer deutsch? ;9 |
| 12:21.45 | NewSole | no deutsch? only english.... tm |
| 12:21.55 | tm | can i asterisk use for conference? |
| 12:21.57 | lahaine | i was stuck the whole day yesterday on odbc not working because of a parsing error .. but without any explicit error message .. just getting .. SQLConnect error message |
| 12:21.57 | tm | NewSole ? |
| 12:22.25 | NewSole | tm yes.... look at MeetMe |
| 12:22.49 | CuTe^G | anyone help me ??? wat is the command to set IP ?? |
| 12:22.57 | NewSole | lahaine> dont feel so bad..... I was in same boat when I first started |
| 12:23.05 | lahaine | hehe :) |
| 12:23.14 | lahaine | anyway, thanks for the tip, it works fine now :) |
| 12:23.17 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 12:23.19 | NewSole | set ip in what CuTe^G> |
| 12:23.28 | NewSole | np lahaine> |
| 12:23.44 | tm | NewSole: meetme? what this? how much peoples? can i use this for 50 peoples and more? assumed i have enough bandwidth? |
| 12:24.10 | NewSole | if you have cpu and bandwith and if they are all using same codec |
| 12:24.47 | tm | NewSole: hm. and what is "MeetMe" |
| 12:25.02 | NewSole | its a conf module for asterisk |
| 12:25.30 | NewSole | but you also need a timing source... like TDM card or a PRI card |
| 12:25.54 | CuTe^G | <PROTECTED> |
| 12:26.12 | CuTe^G | i need to set IP ibn asterisk@home server to open AMP from client computer |
| 12:26.18 | *** join/#asterisk froguz (n=froguz@11-132-222-201.adsl.terra.cl) |
| 12:26.20 | CuTe^G | <PROTECTED> |
| 12:26.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:26.52 | NewSole | I never touched asterisk@home..... I only work with core files... |
| 12:27.12 | MrChimpy | tm: meetme is covered in the book |
| 12:27.15 | *** join/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 12:27.23 | *** part/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 12:27.27 | CuTe^G | ok let me read headbook |
| 12:27.28 | MrChimpy | and a nice book it is too |
| 12:27.30 | CuTe^G | tell me one thing |
| 12:27.38 | CuTe^G | that we can use FAX Modem as a FXO ? |
| 12:27.43 | *** join/#asterisk bofh42 (n=bofh42@p548280B2.dip0.t-ipconnect.de) |
| 12:27.47 | NewSole | nope |
| 12:28.03 | froguz | can i pass commands from the system shell to asterisk CLI? (asterisk < sip show peers) |
| 12:28.05 | tm | MrChimpy: hmm. ok. thx :-9 |
| 12:28.26 | MrChimpy | tm: by that I mean the o'reilly one. i think it's published on web too. |
| 12:28.27 | iDunno | asterisk -rx "sip show peers" |
| 12:28.42 | NewSole | iDunno> you beat me to it... lol |
| 12:28.44 | froguz | thanks iDunno |
| 12:28.50 | MrChimpy | but 50 users in a conference is insanity :) |
| 12:28.57 | iDunno | :) |
| 12:29.21 | NewSole | MrChimpy... no we do all the time |
| 12:29.43 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:30.01 | froguz | CuTe^G, there is an intel modem that can be used as an FXO card (chipset ambient 3200). search for x100p clon in the wiki |
| 12:30.06 | NewSole | but again thats all that box does |
| 12:30.17 | MrChimpy | i suppose if you can organise it like a meeting it'd work. |
| 12:30.34 | *** join/#asterisk NoRemorse (n=bah@202.161.68.2) |
| 12:30.35 | NoRemorse | hi |
| 12:30.37 | MrChimpy | not quite the same for what my place does (chat lines) |
| 12:30.43 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 12:30.50 | NoRemorse | does anyone know how to get g.729 to take dtmf please? |
| 12:31.17 | NewSole | using sip or iax NoRemorse> |
| 12:31.19 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 12:31.28 | NoRemorse | sip |
| 12:31.43 | NewSole | dtmfmode = rfc2833 |
| 12:31.48 | NewSole | dtmf = info |
| 12:31.57 | CuTe^G | Ambient 3200 Intel Modem as a FXO port rite ? |
| 12:32.04 | NewSole | make sure it hose those |
| 12:32.08 | NoRemorse | yeah just dosnt work tho. works with my server (ie voicemail etc) but not with my provider |
| 12:32.36 | NoRemorse | they mentioned something about needing a payload of 100 or something but the asterisk source seems to indicate it adjusts dynamicly |
| 12:32.46 | NewSole | well sounds like provide is only accpting dtmfmode=info |
| 12:32.58 | CuTe^G | ambient modems md3200 |
| 12:33.07 | CuTe^G | ambient modems md3200 we can use for FXO rite |
| 12:33.08 | NoRemorse | yeah well that'll suck. I will ask them tomorrow. |
| 12:33.12 | NewSole | asterisk dtmfpaylod is usaly 101 |
| 12:33.55 | NewSole | this is crazy... not like me to help people... lol |
| 12:34.05 | froguz | CuTe^G, that's right |
| 12:34.21 | NoRemorse | hehe thanks |
| 12:34.29 | *** join/#asterisk MassiveBlue (n=Massive@80.243.63.82) |
| 12:35.14 | NewSole | well cant sit naked all day at computer.... time to take shower and get dressed for work |
| 12:35.22 | NewSole | lol |
| 12:35.27 | NoRemorse | ok just made a call to digium and dtmf works fine |
| 12:39.32 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 12:39.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:41.08 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.217) |
| 12:41.43 | Luhiwu | hello, i'm having some problems with a te110p, is there any support channel? |
| 12:42.37 | trixter | aparently not |
| 12:43.25 | tuxinator_linux | Luhiwu, ask your question |
| 12:44.22 | MassiveBlue | hi there |
| 12:44.38 | vladuk | Q: connection Asterisk <- sipura3000 <- Land Line. who is dealing with DTMF processing spa3000 or Asterisk ? |
| 12:44.57 | MassiveBlue | anybody knows how to install the Asterisk Management Portal on Debian? |
| 12:45.32 | Luhiwu | the E1 line goes down after a few minutes, then up again and down again, and so on... and i have no clue what can be the problem. |
| 12:46.16 | Luhiwu | it looks like electrical problem, but everything is grounded, the E1 comes up and we can pass traffic, then i get red alarm |
| 12:48.44 | CuTe^G | we can use TDM400P all 4 ports as FXO ports or all 4 ports as FXS |
| 12:48.57 | CuTe^G | ? |
| 12:50.04 | MrChimpy | cute: doesn't it rather depend on what modules you have installed in it? |
| 12:50.34 | MatsK | MassiveBlue: Do a search on google and the answer will be releived |
| 12:50.52 | MrChimpy | stick 4 fxo modules in and you have 4 fxo ports. stick 4 fxs in... and you have 4 fxs ports. |
| 12:51.41 | MrChimpy | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 12:51.42 | tuxinator_linux | Luhiwu, I am not familiary with E1's, but I can try to help, any more information that "red alarm" |
| 12:51.48 | MrChimpy | look at the table. it's all there. |
| 12:52.16 | tuxinator_linux | Luhiwu, what is * saying to you? |
| 12:52.36 | MassiveBlue | MatsK: I will try |
| 12:54.07 | fugitivo | i have a box running asterisk 1.0.8 and when the box is rebooted, the sip deviced takes a year to register again, any idea? |
| 12:54.41 | tuxinator_linux | Luhiwu, RED: Loss of signal (LOS): The equipment shall assume "loss of signal" when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. |
| 12:54.47 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:55.03 | wasim | fugitivo: you have set retry timeout to 1 year ... |
| 12:55.13 | fugitivo | wasim: :) |
| 12:56.33 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 12:58.01 | fugitivo | any other idea? |
| 12:59.09 | tuxinator_linux | I'm with wasim on that one |
| 12:59.24 | Luhiwu | tuxinator_linux: i can get the E1 to work ok for a while, then it goes down, very strange |
| 12:59.54 | tuxinator_linux | Luhiwu, it sounds like the signal is poor |
| 13:01.34 | tuxinator_linux | Luhiwu, but I don't know E1's, so don't take my word for it |
| 13:01.41 | fugitivo | wasim: retry timeout where? on the device? |
| 13:02.45 | wasim | Luhiwu: define down ... what does pri debug show? |
| 13:03.00 | wasim | E1 or PRI ? like E1 controlled by SS7 or an ISDN PRI? |
| 13:04.12 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:05.22 | *** join/#asterisk coppice (n=chatzill@33.157.17.210.dyn.pacific.net.hk) |
| 13:06.01 | Luhiwu | wasim: |
| 13:06.03 | Luhiwu | pri show span 1 |
| 13:06.03 | Luhiwu | Primary D-channel: 16 |
| 13:06.03 | Luhiwu | Status: Provisioned, In Alarm, Down, Active |
| 13:06.03 | Luhiwu | Switchtype: EuroISDN |
| 13:06.03 | Luhiwu | Type: CPE |
| 13:06.47 | Luhiwu | e1 isdn pri, bchan 1-15,17-31 dchan 16, as usual |
| 13:08.59 | wasim | something in between the card and far end is down |
| 13:09.31 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:09.38 | wasim | how/when does it come up? by itself? |
| 13:10.15 | Luhiwu | wasim: yes, by itself, after some minutes of plugged in |
| 13:10.36 | wasim | Luhiwu: far end then |
| 13:11.13 | Luhiwu | wasim: the far end says that the problem is my end :-) |
| 13:11.40 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 13:14.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:14.40 | tuxinator_linux | wasim, like wasim said, what does your pri debug show? |
| 13:15.04 | tuxinator_linux | wasim, more specifically, what does it show as when it goes down? |
| 13:15.05 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 13:15.56 | wasim | yeah, wasim |
| 13:17.11 | tuxinator_linux | wasim, sorry, time for bed I guess |
| 13:20.09 | Luhiwu | tuxinator_linux: pri debug shows nothing, i have red alarm in zap show status |
| 13:21.26 | tuxinator_linux | Luhiwu, I assume your verbosity is set very high? |
| 13:21.40 | wasim | and debug to console |
| 13:22.13 | Luhiwu | yes both |
| 13:22.31 | Luhiwu | verbosity is 10, is it high enought? |
| 13:24.46 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 13:28.26 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 13:29.27 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 13:30.03 | *** join/#asterisk The_LightSide (n=dt@wbs-196-2-113-122.wbs.co.za) |
| 13:30.13 | The_LightSide | lo all |
| 13:31.07 | The_LightSide | does anyone know if AAH supports incomming fax detection on bri lines |
| 13:31.25 | tuxinator_linux | Luhiwu, what does it say when you unplug it and plug it back in? |
| 13:31.29 | *** part/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 13:31.48 | tuxinator_linux | Luhiwu, what do they lights on the back say? |
| 13:33.08 | Luhiwu | tuxinator_linux: i'm 60km away now, but yesterday the lights turned green when i plug the cable back in |
| 13:33.23 | *** join/#asterisk DrWho2 (n=MIKE@mike-new.tc3net.com) |
| 13:34.21 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 13:35.27 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 13:35.31 | VJ | Hello |
| 13:36.21 | VJ | i need to configure the sip.conf in such a way that my gatekeeper should receive a call with user name and password defined |
| 13:36.44 | VJ | any hint how to configure that user name and password in sip.conf |
| 13:38.11 | tuxinator_linux | Luhiwu, what does your "span" say in zaptel.conf? |
| 13:38.45 | Luhiwu | tuxinator_linux: span=1,1,0,ccs,hdb3 |
| 13:39.14 | *** join/#asterisk Samoied (n=Samoied@201.24.73.74) |
| 13:39.39 | tuxinator_linux | Luhiwu, do you need "CRC"? |
| 13:41.09 | *** part/#asterisk Samoied (n=Samoied@201.24.73.74) |
| 13:42.18 | RoyK | what is the default echocancel in zap? |
| 13:42.51 | Luhiwu | tuxinator_linux: nop |
| 13:44.28 | tuxinator_linux | Luhiwu, make sure your LBO value is set correctly, the third option |
| 13:44.32 | *** join/#asterisk j0sephi (i=j0e@biztosan.hu) |
| 13:45.24 | tuxinator_linux | Luhiwu, http://www.digium.com/downloads/configuring_zaptel.pdf |
| 13:46.11 | *** join/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 13:46.28 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 13:47.07 | *** join/#asterisk Fr1o (n=Fr1o@AMarseille-252-1-60-25.w86-193.abo.wanadoo.fr) |
| 13:47.11 | Fr1o | hi all |
| 13:47.25 | Luhiwu | tuxinator_linux: tnx, i'll try changing the values |
| 13:48.09 | *** join/#asterisk Abbas (i=Abbas@203.81.196.37) |
| 13:48.10 | santoshr | i get this error when trying to compile mISDNuser ;;;>>>>error: structure has no member named `clone' |
| 13:48.32 | santoshr | i wanted to compile chan_misdn |
| 13:48.50 | santoshr | so got an error for misdnuser files not founf and now this wont install |
| 13:49.30 | *** part/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 13:50.03 | santoshr | any experience with chan_misdn installation ? |
| 13:50.24 | __chris | Hi there - I've got the opportunity to pick up some cheap cisco phones ( 7940 ) - seems to be various licences and so forth - im assuming I don't need any of this to use with asterisk? |
| 13:51.04 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:51.18 | santoshr | <<<error: structure has no member named `clone' >>>> |
| 13:53.20 | tuxinator_linux | __chris, most would recommend that they be running SIP |
| 13:55.14 | [fr10] | hi boys |
| 13:55.44 | [fr10] | I meet lot of difficulties to compil CHAN_BTP |
| 13:55.47 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 13:55.52 | [fr10] | who test it please ? |
| 13:56.18 | santoshr | helllo people.. any ideas with chan_misdn |
| 13:57.05 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 13:57.19 | Katty | mew. |
| 13:57.31 | iDunno | arf. |
| 13:58.18 | [fr10] | somebody try to connect a mobile bluetooth to the pbx ? |
| 13:58.39 | tuxinator_linux | Katty, I stepped on my cat last night, it was dark |
| 13:59.34 | Katty | )= |
| 13:59.51 | tuxinator_linux | Katty, she sure didn't like it |
| 14:00.25 | tuxinator_linux | Katty, I made up for it, I fed her some of my good lunch meat that I was having for my midnight snack |
| 14:00.30 | Katty | well you wouldn't like being stepped on either. |
| 14:00.42 | santoshr | layer.c: In function `mISDNprint_layer_info': |
| 14:00.42 | santoshr | layer.c:201: error: structure has no member named `clone' |
| 14:00.42 | santoshr | layer.c:202: error: structure has no member named `parent' |
| 14:00.43 | wasim | depends if you wore leather and spiky heels |
| 14:00.45 | trixter | I got glow in the dark collars for my cats way back when, when my exfiancee and I split up she kept her cat and I kept mine, mine was smart enough to unbuckle the collar (any collar no matter which one) so I just leave em off now |
| 14:01.05 | trixter | but the glow in the dark ones were the only way we could see em on the floor |
| 14:01.08 | tuxinator_linux | glowing, neat |
| 14:01.16 | Katty | i dont put collars on my kitties. |
| 14:01.24 | Katty | they're all inside cats...they hate the outdoors. |
| 14:01.41 | trixter | yeah /dev/cat is really smart she has unbuckled many leash collars and other stuff.. she also hacked a roommates NT box daily for 2 weeks until she finally got caught in the act |
| 14:01.43 | pif | must one set DYNAMIC_FEATURES in extensions.conf in order to use [featuremap] ? |
| 14:01.44 | tuxinator_linux | mine is any inside also, but just in case she gets out... |
| 14:02.11 | Katty | in case she gets out, you go out after her and bring her back in |
| 14:02.34 | tuxinator_linux | she is kinda fast |
| 14:02.34 | Katty | and in case she gets out, you don't declaw her. |
| 14:02.35 | trixter | Katty: these are inside only cats too, but we needed to be able to see them on the floor. /dev/cat is a brittish blue so she looks all black (cept for a white spot on her belly) and binary her sister was a tortise shell, so they blend in well in low light |
| 14:02.50 | trixter | www.softpaws.com |
| 14:02.57 | tuxinator_linux | I rescued her from the pound, she was already declawed in front |
| 14:03.22 | tuxinator_linux | Mine is a tortise shell |
| 14:03.41 | *** join/#asterisk Asterisk_newbie (n=seymore@203.215.73.192) |
| 14:03.42 | tuxinator_linux | Katty will never be declawed |
| 14:03.54 | Katty | indeed. |
| 14:03.56 | trixter | Katty: www.softpaws.com |
| 14:03.59 | trixter | :) |
| 14:04.14 | trixter | they even come in colors so you can make a statement |
| 14:04.27 | Katty | i wouldn't do that either. |
| 14:04.31 | Katty | something tells me kitty wouldn't like it. |
| 14:04.57 | trixter | they are just covers, any of the cats I have known that have that havent minded once they are on |
| 14:05.14 | Katty | how do yo know? |
| 14:05.19 | Katty | have you asked them? |
| 14:05.19 | trixter | the way htey act |
| 14:05.23 | tuxinator_linux | I wanted to paint my cats claws |
| 14:05.43 | trixter | tuxinator_linux: these look like they are painted if you get the colors |
| 14:05.47 | santoshr | chan_misdn anybody.. |
| 14:05.48 | tuxinator_linux | but I worry about the toxicity |
| 14:05.54 | Asterisk_newbie | any uses voicemaster? |
| 14:06.11 | trixter | Asterisk_newbie: what is voicemaster? sounds kinky |
| 14:06.21 | santoshr | anybody ever in the history of mankind able to install chan_misdn |
| 14:06.24 | Asterisk_newbie | you dont know it? |
| 14:06.32 | Asterisk_newbie | hmmm |
| 14:06.32 | *** join/#asterisk oej (n=oej@199.227.185.35) |
| 14:06.51 | trixter | I might know it just not by that name, what does it do? |
| 14:08.07 | *** part/#asterisk oej (n=oej@199.227.185.35) |
| 14:08.08 | wasim | toxicity is a fun song to play at a concert |
| 14:08.32 | Asterisk_newbie | its a gnugk backend with asterisk feature like sip |
| 14:09.22 | Asterisk_newbie | go to sysmaster.com look for "voipbilling" |
| 14:11.17 | RoadRunnR | santoshr: i'n running it, but with problems |
| 14:11.53 | santoshr | bloddy misdnuser wont install.. |
| 14:12.16 | santoshr | u compiled chan_misdn from channels/misdn |
| 14:12.18 | *** join/#asterisk mjmac (n=mjmac@cpe-69-207-227-173.maine.res.rr.com) |
| 14:13.01 | RoadRunnR | santoshr: please install as described on: http://www.voip-info.org/wiki/view/chan_misdn, this includes everything you need |
| 14:13.03 | santoshr | it must have asked for :"libisdnnet.a " which comes with isdnuser. |
| 14:13.31 | santoshr | i am following exactly those instruciton RoadRunnR |
| 14:13.59 | Luhiwu | anyone knows what does this error means? Write to 44 failed: Unknown error 500 |
| 14:13.59 | santoshr | but am getting stuck. so came here.. but there was so much cat discussion tht my question got lost |
| 14:16.13 | santoshr | i am not able compile misdnuser and i keep getting this error. |
| 14:16.40 | *** join/#asterisk mikimotojdc (n=mikimoto@10.Red-81-36-16.dynamicIP.rima-tde.net) |
| 14:16.51 | mikimotojdc | hello |
| 14:17.01 | santoshr | the page says -------->> install mISDN and mISDNuser: make misdn.. but this thing wont happend// mISDN got done properly kernel patched .. modules ready and loaded.. but misdnuser wont compile.. |
| 14:17.05 | mikimotojdc | is there anyone?? |
| 14:17.17 | mikimotojdc | may i ask one cuestion? |
| 14:17.18 | [fr10] | somebody try to connect a mobile bluetooth to the pbx ? |
| 14:17.20 | *** part/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 14:17.26 | RoadRunnR | santoshr: what error are u getting exactly? |
| 14:17.29 | [fr10] | yes ? |
| 14:17.46 | santoshr | wait i will paste |
| 14:17.54 | [fr10] | yes ? |
| 14:17.54 | docelm0 | Not here.. |
| 14:17.58 | docelm0 | use pastebin |
| 14:18.06 | tm | NewSole: hello? |
| 14:18.12 | [fr10] | yes ? |
| 14:18.13 | santoshr | yeah i know dude.. thts why i told him to hold on |
| 14:18.16 | tm | Meaty: is your meetme compiled? |
| 14:18.19 | docelm0 | [fr10], SHUT UP! |
| 14:18.19 | tm | aeh |
| 14:18.26 | tm | NewSole: is your meetme compiled? |
| 14:18.42 | docelm0 | [fr10], yes I have messed with it and was unsuccessful bluetooth on asterisk is still fairly buggy |
| 14:18.53 | [fr10] | docelm0 , i think you don't test bluetooth :-) |
| 14:18.59 | [fr10] | ah ok |
| 14:19.05 | [fr10] | can we talk in private ? |
| 14:19.27 | docelm0 | [fr10], sigh.. no time.. I just pop in and out.. Im at work right now.. |
| 14:19.42 | [fr10] | so SHUT UP please |
| 14:19.45 | [fr10] | ;) |
| 14:19.49 | [TK]D-Fender | ~pb |
| 14:19.51 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:20.32 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:20.34 | santoshr | RoadRunnR: .. please check this out http://sial.org/pbot/15480. |
| 14:20.46 | Meaty | tm : ;) You awake me ! |
| 14:20.48 | docelm0 | sigh your the one being annoying.. with your yes ? yes ? yes ? geesh.. |
| 14:20.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.09 | tm | Meaty: hey, sorry. :-9 |
| 14:21.10 | RoadRunnR | santoshr: wrong mISDN package |
| 14:21.18 | mikimotojdc | i'm planning to build a carrier grade system with asterisk. The problem is this (i'm new in this system): when a call cam to the system i must know if the CLI number is activated in my system and if the client has credit. The problem is that when the call is coming i can not hang on the call. first i must check than an place another call an then y can hang on |
| 14:21.20 | santoshr | meaning ? |
| 14:21.23 | mikimotojdc | so |
| 14:21.35 | jimmy_deanPB | I'm having some issues with a zap channel in a TDM400P going off hook after 30 seconds or so of plugging a line into it. The other channel with the other line does not go off hook...is this a common problem that can be solved with certain settings in zapata.conf? |
| 14:21.35 | mikimotojdc | it is posible to do it with asterisk?? |
| 14:21.41 | RoadRunnR | santoshr: in the svn checkout path, got to channels/misdn and do the make misdn there |
| 14:22.09 | santoshr | i tried doping tht dude.. after make in wont install cuz its not able to find some files .. |
| 14:22.18 | santoshr | and those files belong to misdnuser |
| 14:22.51 | santoshr | libisdnnet.a: and this file would come once i can make misdnuser.. |
| 14:23.00 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 14:23.26 | *** join/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 14:24.00 | jimmy_deanPB | hey [TK]D-Fender, how's it going? |
| 14:24.06 | RoadRunnR | the libisdnnet.a file will be create by a make or make all in the channels/misdn directory after the make misdn |
| 14:24.13 | froguz | is it possible to add a sip peer to sip.conf throw the CLI (like add extension)? |
| 14:24.26 | santoshr | but it says no rule.. it doesnt understand make misdn |
| 14:24.26 | [TK]D-Fender | jimmy_deanPB : NTB ATM |
| 14:24.29 | [fr10] | can you help me using chan_btp please ? |
| 14:24.34 | santoshr | i have asterisk 1.2.0 |
| 14:24.40 | [fr10] | if somebody know what is it |
| 14:24.41 | [fr10] | :) |
| 14:25.05 | jimmy_deanPB | [TK]D-Fender, I think our Digium hardware is giving us problems |
| 14:25.17 | santoshr | RoadRunnR: i hvae 1.2.0 and it wont recognize make misdn |
| 14:25.24 | pif | in ordre to use [featuremap] must one declare them in extensions.conf ? |
| 14:25.28 | RoyK | ~seen zoa |
| 14:25.42 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 4d 17h 25m 51s ago, saying: 'apt-get install libssl-dev'. |
| 14:25.51 | RoyK | santiago: rule number one: Always run the latest release |
| 14:26.01 | RoyK | ~rules |
| 14:26.03 | jbot | methinks rules is Ktron makes the rules! |
| 14:26.16 | RoyK | oh |
| 14:26.16 | [TK]D-Fender | jimmy_deanPB : So I read. Is that only the time that you plug the lin onto it physically? |
| 14:26.17 | VJ | <PROTECTED> |
| 14:26.27 | RoadRunnR | santoshr: i never got 1.2.0 to run properly |
| 14:27.03 | santoshr | which one are u using |
| 14:27.03 | VJ | i need to send a call to the PSTN provider via VOIP, but the call needs to be registered via user name and password they hve provided me |
| 14:27.28 | jimmy_deanPB | [TK]D-Fender, yeah, once any of our analog lines are plugged into that Zap channel, 30 seconds later you hear the operator saying "If you'd like to make a call, please hang up and try again..." and then the off-hook sound |
| 14:27.43 | Luhiwu | any pri expert here? i'm getting unknown error 500 |
| 14:27.54 | [TK]D-Fender | jimmy_deanPB : "Plugged into that zap channel" ? huh>? |
| 14:28.09 | jimmy_deanPB | [TK]D-Fender, Zap channel #4 |
| 14:28.15 | jimmy_deanPB | on the TDM400P |
| 14:28.42 | [fr10] | can you help me using chan_btp please ? |
| 14:29.06 | VJ | i need to send a call to the PSTN provider via VOIP, but the call needs to be registered via user name and password they hve provided me |
| 14:29.13 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 14:29.17 | VJ | any idea how to configure sip.conf for that |
| 14:29.24 | anderiv | [fr10]: you've asked plenty of times. Aparrently no one can help you here. Please move on. |
| 14:29.58 | [TK]D-Fender | jimmy_deanPB : You mean only once at the time you physically plug it in? |
| 14:30.06 | santoshr | RoadRunnR: which asterisk are u using |
| 14:30.06 | jimmy_deanPB | [TK]D-Fender, ztmonitor shows Zap/3 with one bar of sound level but Zap/4 has zero bars...you can also hear a difference in volume if you connect an analog phone |
| 14:30.11 | brad_mssw | recommended voip providers for US (unlimited simultaneous lines) ?? |
| 14:30.28 | RoadRunnR | santoshr: the latest from the svn crichter branch |
| 14:30.46 | jimmy_deanPB | [TK]D-Fender, yeah, then it stays off hook until you plug the analog line straight into an analog phone and hangup |
| 14:31.07 | RoadRunnR | santoshr: however, this version causes problems with my HFC card :-( |
| 14:31.09 | [TK]D-Fender | jimmy_deanPB : Not sure what to tell you... |
| 14:31.19 | jimmy_deanPB | [TK]D-Fender, that's ok, I'm going to call Digium on it |
| 14:31.24 | [TK]D-Fender | Good idea... |
| 14:31.30 | jimmy_deanPB | I thought I'd ask in here first though |
| 14:31.41 | santoshr | so.. wht are u trhin to say.. wht are people over here saying. |
| 14:31.50 | santoshr | is it supposed to behave this way |
| 14:32.27 | jimmy_deanPB | [TK]D-Fender, the new dialplan is working great by the way |
| 14:32.29 | santoshr | but what i am thinkin is htt once i can get misdnuser compiled .. everythion will work |
| 14:32.30 | santoshr | .. |
| 14:32.39 | [fr10] | can you help me using chan_btp please ? |
| 14:33.26 | [TK]D-Fender | jimmy_deanPB : Good to hear. I've solidified on 2 major plan styles, and have gotten my own home up and running 100% |
| 14:33.37 | *** join/#asterisk BeautifulMess (n=Shalom@dslb-084-063-106-248.pools.arcor-ip.net) |
| 14:34.32 | RoadRunnR | santoshr: my experience is that misdn can extremly version depended, i had one version that was rock stable, then i upgrade to 1.2.0 and nothing worked anymore, the new 0.3.0 stuff is beta and has some problems still, but is overall quite promising so far |
| 14:34.43 | *** join/#asterisk foucaulo (n=fouca@modemcable186.159-82-70.mc.videotron.ca) |
| 14:35.03 | jimmy_deanPB | [TK]D-Fender, nice! I have to reinstall Debian on my home box since I did an upgrade to Debian unstable to try out their new asterisk packages but decided that sucked and went back down to stable, but that didn't over very well |
| 14:35.25 | santoshr | what are others sayin.. are we the only two trying to get chan_misdn to work ? |
| 14:35.27 | Luhiwu | any pri expert here? i'm getting unknown error 500 |
| 14:35.40 | [TK]D-Fender | jimmy_deanPB : Never install * from packages.... it just sucks. Always use source for it and You won't have to wory about the rest of your system. |
| 14:36.18 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 14:36.36 | cypromis | hi little red riding hood |
| 14:36.37 | cypromis | :P |
| 14:37.20 | *** part/#asterisk BeautifulMess (n=Shalom@dslb-084-063-106-248.pools.arcor-ip.net) |
| 14:37.41 | [TK]D-Fender | cypromis : RedHat :) |
| 14:38.10 | [TK]D-Fender | et pis salut mon ostie! ;) |
| 14:38.21 | cypromis | yah I know |
| 14:38.42 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 14:38.51 | zoa | hey ho |
| 14:38.59 | [fr10] | can you help me using chan_btp please ? |
| 14:39.20 | jero | btp? batimen travaux publics |
| 14:39.27 | santoshr | RoadRunnR: just got to comple misdnuser properly.. compiling asrtierk again.. |
| 14:39.36 | jimmy_deanPB | [TK]D-Fender, oh I know, that was a while ago...I've learned first hand the value of compiling asterisk by hand :) |
| 14:39.46 | [TK]D-Fender | jero : Bluetooth |
| 14:39.59 | [TK]D-Fender | jimmy_deanPB : Yeah.. it tends to work :) |
| 14:40.02 | jero | :) |
| 14:40.36 | [fr10] | you test asterisk with a mobile bluetooth? |
| 14:42.14 | foucaulo | somebody can help me? i have calls dropping with channels between 2 box at each 640 seconds...(it do the same thing in sip and IAX2) |
| 14:42.34 | foucaulo | (2 asterisk box) |
| 14:42.59 | *** join/#asterisk BlueMassive (n=Massive@80.243.63.82) |
| 14:43.10 | [TK]D-Fender | jero : J'ai trouve la facture que je vous-doit (TDM22B). Je comprends pas je l'ai perdu.... t'en veut tu que je te le fax? |
| 14:44.19 | [fr10] | bon ils sont où les gaulois? |
| 14:44.23 | [fr10] | et les gauloises ? |
| 14:44.27 | [fr10] | :) |
| 14:44.33 | *** join/#asterisk viperdude (n=jon@borat.enta.net) |
| 14:44.49 | pif | [fr10] : you have to speak french? |
| 14:46.40 | foucaulo | someone can tell to me what is the best way to get help here?----Est-ce que quelqu'un peut me dire quelle est la meilleur façon d'obtenir de l'aide ici? |
| 14:46.57 | [fr10] | speak on the chan |
| 14:46.59 | [fr10] | ;) |
| 14:47.02 | fugitivo | foucaulo: just ask your question in english and wait |
| 14:47.11 | [TK]D-Fender | foucaulo : Try asking without being repeitive and annoying :) What do you need help with? |
| 14:47.21 | anderiv | [TK]D-Fender: heh - well put :-) |
| 14:48.20 | [TK]D-Fender | [fr10] : This means YOU. Chan_bluetooth is experimental and VERY few people have tried it and fewer are online HERE. Try asking every other HOUR instead of every other MINUTE. |
| 14:48.22 | foucaulo | [TK]D-Fender: i need help to know if i'm wrong or if i find a bug in asterisk |
| 14:48.42 | [TK]D-Fender | foucaulo : Ok, now try telling us the actual problem :) |
| 14:48.47 | fugitivo | lol |
| 14:48.54 | [TK]D-Fender | Getting warmer.... |
| 14:49.54 | fugitivo | ... |
| 14:50.28 | foucaulo | [TK]D-Fender: I'm using one sip phone who register to an Asterisk box1. This box1 place calls made by the sip phone to the asterisk box 2 with iax2 channel ...... |
| 14:50.29 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:50.43 | iCEBrkr | shit fuck damn piss |
| 14:50.46 | *** join/#asterisk IRS (n=Nbm@87-196-8-26.net.novis.pt) |
| 14:50.50 | foucaulo | [TK]D-Fender: all call are drop at 640 sec |
| 14:50.59 | iCEBrkr | ow! |
| 14:51.04 | [TK]D-Fender | Watch your language! |
| 14:51.13 | iCEBrkr | What's it supposed to do? Tricks? |
| 14:51.23 | foucaulo | [TK]D-Fender: I have also try in sip and it do the same thing |
| 14:51.27 | [TK]D-Fender | foucaulo : exactly taht time? |
| 14:51.36 | iCEBrkr | Corydon's ODBC patch is leet and all, but I can't do multiple databases without setting up another ODBC DSN |
| 14:51.42 | iCEBrkr | :( |
| 14:51.55 | [TK]D-Fender | foucaulo : Pastebin your extensions.conf |
| 14:51.55 | *** join/#asterisk svenna_ (n=svenna@p548D1EDB.dip0.t-ipconnect.de) |
| 14:51.56 | [TK]D-Fender | ~pb |
| 14:52.00 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:52.12 | foucaulo | [TK]D-Fender it may change a little bit but always 10minutes 30 sec |
| 14:52.16 | fugitivo | foucaulo: what's the sip phone? |
| 14:53.13 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 14:53.34 | iCEBrkr | Hrrmm.. If I set a variable in a call file to something like "SELECT * FROM ${TABLE}}" will the ${TABLE} eval once it's in Asterisk? :P |
| 14:53.37 | foucaulo | fugitivo: it's a Gentek but it's made by Azatell ....... iknow that's not the phone ... y try with other other way..... |
| 14:53.53 | [TK]D-Fender | foucaulo : ok, copy your extensions.conf into www.pastebin.com and let us take a look |
| 14:54.07 | foucaulo | [TK]D-Fender: how to do that? |
| 14:54.17 | *** join/#asterisk tengulre (n=tengulre@61.150.60.192) |
| 14:54.20 | [TK]D-Fender | iCEBrkr : Is it being called like that from the dialplan? |
| 14:54.35 | iCEBrkr | Not yet. I'm just wondering |
| 14:54.39 | fugitivo | tengulre [n=tengulre@61.150.60.192] has joined #ASTERISK |
| 14:54.47 | fugitivo | why is #ASTERISK in uppercase? |
| 14:54.47 | [TK]D-Fender | iCEBrkr : If its in the dial-plan it should eval. |
| 14:54.48 | fugitivo | weird |
| 14:54.50 | tengulre | :) |
| 14:54.56 | iCEBrkr | SetVar: _SQL = "SELECT * FROM ${TABLE}" |
| 14:55.06 | iCEBrkr | So then in the dialplan I'd getch ${SQL} |
| 14:55.26 | iCEBrkr | s/getch/fetch |
| 14:55.40 | DrWho17 | hrm, just tried voxee.com, they sure are slow at getting the call out |
| 14:55.42 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 14:55.53 | *** part/#asterisk tengulre (n=tengulre@61.150.60.192) |
| 14:56.02 | *** join/#asterisk tengulre (n=tengulre@61.150.60.192) |
| 14:56.34 | tengulre | Is it OK? |
| 14:56.35 | *** part/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:56.42 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:56.44 | [TK]D-Fender | :O |
| 14:56.50 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 14:56.52 | tengulre | o<|:O) |
| 14:57.23 | [TK]D-Fender | *I*? |
| 14:57.29 | foucaulo | [TK]D-Fender: ------>http://pastebin.com/509907 |
| 14:57.36 | Katty | mister fender. |
| 14:57.48 | [TK]D-Fender | Ms. Katty: ...... mew. |
| 14:58.07 | Katty | ewwo. |
| 14:58.41 | [TK]D-Fender | foucaulo : ok, now pastebin the IAX.CONF entry for that. |
| 14:59.08 | [TK]D-Fender | Katty: Owareou? |
| 14:59.20 | Katty | sleepy. |
| 14:59.41 | [TK]D-Fender | Katnap time.... |
| 15:00.01 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 15:01.11 | foucaulo | [TK]D-Fender: http://pastebin.com/509910 |
| 15:01.18 | Katty | [TK]D-Fender: shan't...i have work stuffs to do. |
| 15:02.25 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 15:03.52 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 15:04.20 | foucaulo | [TK]D-Fender: possibly good to know....: the box2 place calls to a sip VoIP provider |
| 15:04.41 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-134.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:04.46 | [TK]D-Fender | foucaulo : have you tested a call to just an extension on Box2 and timed it? |
| 15:04.49 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 15:05.51 | *** part/#asterisk bkw_ (n=bkw_@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net) |
| 15:07.09 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:07.25 | foucaulo | [TK]D-Fender: It test it right now... I also know that it's the box1 that hangup the call.... cause code 16 |
| 15:08.14 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:08.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:09.04 | *** part/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 15:09.04 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 15:09.20 | *** join/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 15:09.46 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net) |
| 15:11.16 | *** join/#asterisk smurfix (n=smurf@debian/developer/smurf) |
| 15:11.43 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 15:12.27 | Trazz | can you have all the ivr and prompts be in the womans voice from the demo and if so how? |
| 15:12.41 | DrWho17 | hire her |
| 15:13.40 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:14.01 | Trazz | hehe nice.. is there a way or not? |
| 15:14.13 | Trazz | it would seem weird to havfe her voice and then mine in there |
| 15:14.14 | [TK]D-Fender | Trazz : Go to http://www.theivrvoice.com/ |
| 15:14.57 | DrWho17 | http://www.theivrvoice.com/html/testimony.htm |
| 15:16.14 | foucaulo | [TK]D-Fender : may it be a bug similar to http://bugs.digium.com/view.php?id=3582&nbn=11 ?? |
| 15:16.18 | Trazz | how much is she?? |
| 15:16.25 | zoa | Trazz: prices are on the website |
| 15:16.31 | DrWho17 | reasonable |
| 15:16.50 | [fr10] | can you help me using chan_btp please ? |
| 15:21.38 | RoyK | zoa: wb |
| 15:21.47 | zoa | hey ho |
| 15:21.59 | zoa | i asked abou this crash you had |
| 15:22.03 | zoa | he cannot fix it |
| 15:22.11 | zoa | he can make sure it doesnt crash |
| 15:22.11 | jimmy_deanPB | [TK]D-Fender, aha, self inflicted pain...we think we solved the issue...a bad phone cord going from our patch panel to the TDM400P |
| 15:22.12 | RoyK | oh |
| 15:22.15 | zoa | but then you will not have audio |
| 15:22.20 | [TK]D-Fender | jimmy_deanPB :O |
| 15:22.21 | zoa | on this call |
| 15:22.21 | RoyK | :P |
| 15:22.31 | RoyK | what is wrong |
| 15:22.32 | zoa | he actually forces it to crash there |
| 15:22.32 | RoyK | ? |
| 15:22.35 | RoyK | zoa: did you see my note on mantis? |
| 15:22.38 | jimmy_deanPB | [TK]D-Fender, upgraded to a nice CAT5e cable |
| 15:22.42 | RoyK | i know |
| 15:22.44 | RoyK | i saw it |
| 15:22.46 | zoa | yes, it is zap -> iax2 |
| 15:22.51 | RoyK | but why does he do so? |
| 15:22.57 | [TK]D-Fender | jimmy_deanPB : You plugged CAT into the wall jack? |
| 15:23.01 | [TK]D-Fender | CAT5* |
| 15:23.04 | zoa | because there is something wrong at that point |
| 15:23.15 | zoa | anyhow, he could probably fix all that, but is going to take much much more time |
| 15:23.23 | Trazz | yep she is $150.00 per hour which is reasonable for her quality voice |
| 15:23.28 | jimmy_deanPB | [TK]D-Fender, wait, what? |
| 15:23.37 | Trazz | just talked to Allison |
| 15:23.40 | Trazz | very nie |
| 15:23.43 | Trazz | very nice |
| 15:23.49 | RoyK | zoa: but if you don't support iax jb, why did you modify configs/iax.conf-sample? |
| 15:23.49 | zoa | say hi from zoa and dave |
| 15:23.50 | zoa | :) |
| 15:23.57 | Trazz | thanks for the tip guys |
| 15:24.04 | zoa | because we made this one especially for you |
| 15:24.14 | zoa | to try it without having to make too much changes |
| 15:24.23 | Trazz | i talked to her about doing exclusive with me so i can then be her agent |
| 15:24.37 | [TK]D-Fender | jimmy_deanPB : You said you plugged CA5e betweent the TDM and the wall... just wondering how your wall jack could accept that |
| 15:24.38 | zoa | it works with the other patch you had, but its crashing |
| 15:24.38 | Trazz | : |
| 15:24.50 | Trazz | we can take that rate to $500 per hour i told her :) |
| 15:24.57 | Trazz | she wants to see the contract now |
| 15:24.58 | Trazz | heheeheeh |
| 15:25.11 | jimmy_deanPB | [TK]D-Fender, oh, it's a universal jack |
| 15:25.25 | jimmy_deanPB | [TK]D-Fender, it'll accept RJ45 and whatever the phone cord connector is |
| 15:25.39 | [TK]D-Fender | jimmy_deanPB : interesting... |
| 15:26.01 | jimmy_deanPB | yeah, makes things really simple |
| 15:26.24 | jimmy_deanPB | we also have universal patch panels too, you can hook up analog phone or ethernet |
| 15:29.07 | foucaulo | [TK]D-Fender: when a call an extension of box2, call is not dropping....... |
| 15:29.17 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 15:29.30 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 15:29.46 | hickins | hello room, I've much trouble finding this room |
| 15:30.02 | Trazz | well the door wasn't locked |
| 15:30.24 | hickins | can anyone help me with iax client connection problem? |
| 15:30.46 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc12c.dialup.mindspring.com) |
| 15:30.49 | [TK]D-Fender | foucaulo : Could be a problem with the voip provider |
| 15:32.06 | hickins | I'm not an expert in chat rooms like these, can I start telling my problem? |
| 15:32.17 | tuxinator_linux | hickins, yes |
| 15:32.20 | hickins | do I have a listner? |
| 15:32.28 | tuxinator_linux | hickins, maybe |
| 15:32.31 | hickins | thanks |
| 15:32.44 | file | you're a listener! |
| 15:32.56 | tuxinator_linux | hickins, you ask, and if someone knows, they answer, otherwise, try again later |
| 15:33.00 | hickins | well I connect to * server through a vpn tunnel |
| 15:33.17 | foucaulo | [TK]D-Fender: the reason the i don't think so is because when a place a call to the voip provider on the box1 and the sip phone, in the both ways, call dont drop.... |
| 15:33.42 | tuxinator_linux | file, don't tell my wife ;-) |
| 15:33.50 | hickins | so when I tried to setup an iax phone (software) it says it rejects the connection |
| 15:33.59 | hickins | in the * cli window |
| 15:34.20 | *** join/#asterisk ckruetze (n=ckruetze@i577A01D6.versanet.de) |
| 15:34.32 | hickins | in iax.conf I've set the bindingaddr to 0.0.0.0 |
| 15:34.38 | tuxinator_linux | hickins, pastebin your exact error message |
| 15:34.42 | tuxinator_linux | ~pb |
| 15:34.43 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:35.03 | hickins | and I have set an iax account as a freind |
| 15:35.15 | hickins | with a user name and a secre |
| 15:35.22 | tuxinator_linux | nope |
| 15:35.27 | tuxinator_linux | you can "XXXXX" them out |
| 15:35.36 | *** join/#asterisk ManxPower (i=ewieling@217.sub-70-197-249.myvzw.com) |
| 15:35.47 | tuxinator_linux | hickins, oops, read it wrong, pleaes continue |
| 15:36.11 | ckruetze | Hi, I'm having some trouble with our firewall and SIP. Could someone try to give me a testcall to christian@balticfinance.com |
| 15:36.22 | hickins | Jan 17 18:33:19 NOTICE[2963]: chan_iax2.c:5448 socket_read: Rejected connect attempt from 172.31.1.3 |
| 15:36.40 | hickins | any suggestions? |
| 15:36.55 | tuxinator_linux | hickins, I'm sorry, I'm a little too sleepy right now, maybe someone else can help |
| 15:37.17 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 15:38.02 | foucaulo | [TK]D-Fender: how we can explain that? |
| 15:38.18 | jpkoopmann | ckruetze: Now? |
| 15:38.28 | [TK]D-Fender | foucaulo : Do calls from Box2 to the SIP provider work fine? |
| 15:38.37 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:39.13 | Luhiwu | anyone knows what does this means? |
| 15:39.13 | Luhiwu | Write to 43 failed: Unknown error 500 |
| 15:39.14 | Luhiwu | Short write: 0/15 (Unknown error 500) |
| 15:39.22 | hickins | I even tried remark bindaddr so that it will accept connections from everywhere |
| 15:39.55 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 15:39.59 | ckruetze | jpkoopmann: I think it is not working on our side :( I'll do some more work and come back with the request |
| 15:40.00 | hickins | of course I did not forget to reload everytime I change |
| 15:40.51 | foucaulo | [TK]D-Fender: configuration used is the same that I tested on box1 (result where great..) to be sure I test it right now....... |
| 15:40.52 | file | hickins: it couldn't authenticate, that would be the simplest answer |
| 15:41.24 | *** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com) |
| 15:50.23 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 15:55.28 | Trazz | does * do ACD ? |
| 15:55.58 | fugitivo | yes |
| 15:56.41 | *** join/#asterisk lahaine (n=lahaine@63.66.119-80.rev.gaoland.net) |
| 15:57.38 | *** join/#asterisk SYS64738 (n=giaco@host230-254.pool81123.interbusiness.it) |
| 15:58.23 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 16:01.04 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 16:01.36 | SYS64738 | ciao |
| 16:01.46 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 16:02.03 | MrChimpy | blimey. that's C64 stuff isn't it? |
| 16:02.56 | RoyK | hm... |
| 16:03.11 | RoyK | is there a way to split a string in a variable in the dialplan without using agi? |
| 16:03.31 | RoyK | like in perl ($a,$b) = split(/\./, $c); |
| 16:03.37 | SYS64738 | yes is the soft reset |
| 16:04.17 | [TK]D-Fender | RoyK : if ${a} = abcdefg then ${a:2:3} = cde. |
| 16:04.21 | wasim | RoyK:4:-3 |
| 16:04.22 | [TK]D-Fender | RoyK : Like that? |
| 16:04.27 | MrChimpy | ah. only 15 years since I went near C64 :) |
| 16:04.32 | SYS64738 | poke53280,0 poke53281,0 poke 646,7 |
| 16:04.52 | anthm | show function cut |
| 16:05.03 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.103.247) |
| 16:05.06 | RoyK | wasim: it's a number so the number of digits may vary |
| 16:05.17 | RoyK | wasim: and asterisk saynumber can't do decimals :P |
| 16:06.07 | [TK]D-Fender | RoyK : There is a way.... |
| 16:06.07 | m_a_g_o | hi, * is detecting also inband tones besides RFC2833 RTP Events, what can I do? this really represents a problem! |
| 16:06.08 | RoyK | [TK]D-Fender: there usually is |
| 16:06.21 | tzanger | m_a_g_o: have the far end pick one or the toher, not both. |
| 16:06.21 | RoyK | [TK]D-Fender: only i do not know |
| 16:06.43 | [TK]D-Fender | RoyK : I believe you can check for the location of "." and play back a recording for it, then use say digits for the remainder. |
| 16:07.14 | [TK]D-Fender | RoyK : So saynumber (whole) + "point" + Saydigits(decimal) |
| 16:07.26 | twisted[asteria] | show function cut |
| 16:07.38 | twisted[asteria] | 'das all I'm saying ;) |
| 16:08.13 | m_a_g_o | both ends have RFC2833, but the problem seems to be a feedback of the tones phones reproduces in the head set, and Asterisk seems to get them, because I only see the correct amount of RTP Events... besides that only payload |
| 16:09.23 | twisted[asteria] | anthm, heh. |
| 16:09.24 | ManxPower | m_a_g_o, the phone should not be sending inband dtmf if it's set for OOB DTMF. |
| 16:09.42 | *** part/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 16:10.12 | ManxPower | hickins, you have a mismatch between what is configured in iax.conf and what the remote side is sending |
| 16:10.39 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 16:10.39 | m_a_g_o | This the call flow: Cell Phone -> PSTN -> VoIP provider (RF2833) -> * (RFC2833) |
| 16:11.01 | Biappi | an ISDN phone can send OOB dtmf? |
| 16:11.19 | wasim | Biappi: by design |
| 16:11.44 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 16:11.59 | Biappi | uhm.. so i've not understood a thing |
| 16:12.37 | Biappi | even if i have oob dtmf i hear them during the call? |
| 16:12.48 | Biappi | or maybe the phone is sending both? |
| 16:13.35 | ManxPower | I thought only send inband DTMF (as you would see if you did a PRI or BRI debug and generated DTMF). |
| 16:13.38 | wasim | or your phone says, lookeee a DTMF signal in the d channel, lemmo confuse Biappi |
| 16:13.48 | ManxPower | Now DIALING is OOB, but that's just a message, not DTMF |
| 16:14.04 | m_a_g_o | anyway, any ideas on how to configure asterisk's sensitivity or how to tell asterisk not to receiva inband tones if it is configured to use RFC2833? |
| 16:14.16 | Math` | dtmfmode=rfc2833 ... |
| 16:14.27 | m_a_g_o | already done |
| 16:14.48 | ManxPower | m_a_g_o, In theory it should NOT listen for inband DTMF if it's configured for RFC2833, but I suspect it does so anyway. You would have to look at the code to be sure. |
| 16:15.15 | ManxPower | m_a_g_o, BTW, are you SURE the provider is using RFC2833? Also what codec are you using? |
| 16:15.32 | m_a_g_o | yes, I'm seeing the packets in tethereal |
| 16:15.43 | ManxPower | m_a_g_o, Why not use IAX to the provider? Do they not support it? |
| 16:16.08 | ManxPower | m_a_g_o, So is the provider passing the inband DTMF from the cell phone? |
| 16:16.14 | m_a_g_o | no... unfortunately... |
| 16:16.35 | m_a_g_o | yes, it appears to be passed included with the audio |
| 16:16.48 | ManxPower | <PROTECTED> |
| 16:16.51 | m_a_g_o | as an audible tone |
| 16:17.09 | ManxPower | m_a_g_o, If you can prove it, report it as a bug. |
| 16:17.50 | m_a_g_o | yes... that is clearly the problem, if dialing from a PSTN analog phone and setting it in MUTE while dialing the tones, it works just fine. |
| 16:18.09 | m_a_g_o | not MUTE, duplicate tones |
| 16:18.14 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 16:18.24 | ManxPower | m_a_g_o, is sounds like your provider is doing this, not asterisk. |
| 16:18.43 | ManxPower | i.e. the provider is sending both the inband DTMF and the RFC2833 DTMF. |
| 16:19.05 | twisted[asteria] | zomg |
| 16:19.08 | ManxPower | m_a_g_o, Anyway, report it as a bug. The mute/nomute is a critical thing in this report. |
| 16:19.16 | twisted[asteria] | it's called you have your phone turned up too loud. |
| 16:19.20 | ManxPower | m_a_g_o, also what version of Asterisk? |
| 16:19.22 | m_a_g_o | how can you not send an audible tone the phone picks up? you would have to filter that frequency, but you can't since it is audio! |
| 16:20.02 | ManxPower | m_a_g_o, yes, but your PROVIDER should be not passing those tones on to Asterisk. |
| 16:20.20 | m_a_g_o | it doesn't, I see the correct amount of RTP Events |
| 16:20.26 | ManxPower | I suspect Asterisk just forgets to turn off inband dtmf detection when dtmfmode=rfc2833 is set |
| 16:20.47 | ManxPower | m_a_g_o, But you also get audio send to Asterisk, so you are getting both sent to Asterisk |
| 16:20.57 | m_a_g_o | yes |
| 16:21.19 | m_a_g_o | where can I post the possible bug? |
| 16:21.30 | ManxPower | the device that does the PSTN/VoIP conversion should be the device that filters out inband dtmf |
| 16:21.34 | ManxPower | m_a_g_o, bugs.digium.com |
| 16:21.39 | twisted[asteria] | NO |
| 16:21.48 | twisted[asteria] | there is already a bug there it seems. |
| 16:22.04 | twisted[asteria] | see bug id 6027 |
| 16:22.08 | m_a_g_o | I believe you can't filter them since they are part of the audio... |
| 16:22.11 | ManxPower | twisted[asteria], Oh, I agree it sounds like a bug, but it sounds like a bug in BOTH the provider and in Asterisk. |
| 16:22.44 | ManxPower | m_a_g_o, many devices that do PSTN/VoIP conversion just MUTE all audio for the duration of the DTMF |
| 16:23.17 | m_a_g_o | hmmm, Ok I see, it would be nice to have one right now... |
| 16:23.19 | ManxPower | Argh! With all this rain I'm beginning to wonder if I should start building an ark. |
| 16:23.22 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 16:23.46 | twisted[asteria] | ManxPower, now would not be the time |
| 16:24.01 | twisted[asteria] | ManxPower, by the time you're finished, it'll be gone :) |
| 16:25.02 | argentas | anyone know what Digium's VoiceBus technology is/does? (as in "TDM2400P with new VoiceBus Technology") |
| 16:25.32 | m_a_g_o | ManxPower, isn't there a way to adjust Asterisk's length parameter to recognize a tone? |
| 16:25.45 | ManxPower | m_a_g_o, no. |
| 16:26.19 | ManxPower | m_a_g_o, you can adjust the SENDING DTMF tone length, but for receiving it, well that's voodo and nobody understands it except for Mark. |
| 16:26.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:26.33 | ManxPower | I think he sacraficed two chickens and a goat to make it work. |
| 16:27.05 | argentas | I thought it was two goats and a chicken, but i may be wrong.. ;-) |
| 16:27.09 | tzanger | the receiving is not that difficult ot understand |
| 16:27.16 | tzanger | it's all in dsp.c and actually not badly documented |
| 16:27.21 | m_a_g_o | jajaja, ok... I may have to go out of town to find live animals though... |
| 16:28.11 | ManxPower | tzanger, vonunteering to fix it, I see? |
| 16:28.17 | tzanger | ManxPower: what's wrong with it? |
| 16:28.59 | ManxPower | tzanger, Asterisk detects DTMF sent inband, even when it's set for RFC2833 |
| 16:29.22 | ManxPower | twisted[asteria] see bug id 6027 |
| 16:29.32 | tzanger | ManxPower: I thought you were talking about lengthening the time it takes to detect DTMF |
| 16:29.45 | coppice | ManxPower: most things do that |
| 16:29.50 | ManxPower | tzanger, nope. |
| 16:29.55 | jbalcomb | can I adjust the verbosity level of logging while asterisk is running? |
| 16:29.58 | twisted[asteria] | ManxPower, i said at 10:22 to see that bug id. |
| 16:30.09 | twisted[asteria] | ManxPower, so I'm one step ahead of ya there ;) |
| 16:30.20 | *** join/#asterisk kryl (n=gui@217.167.228.105) |
| 16:30.20 | tzanger | I don't *think* it'd be that difficult to turn off the DTMF detector in dsp.c for that leg of the call, but I haven't looked into it |
| 16:30.22 | kryl | hi |
| 16:30.28 | ManxPower | coppice, *shrug* maybe so, but it still causes problems for people when talking to devices that send both inband and rfc2833 dtmf at the same time |
| 16:30.31 | tzanger | actually is 't' or 'T' being used? |
| 16:30.33 | kryl | Is there a way to simply order to call a numero ? |
| 16:30.46 | kryl | I just want to test if my fxo card is well connected to the line |
| 16:30.55 | coppice | ManxPower: if a device sends both its seriously broken |
| 16:30.56 | m_a_g_o | twisted[asteria]: that bug seems to be an RTP problem, all RTP are corrct in this case |
| 16:31.04 | kryl | x100p |
| 16:31.10 | tzanger | m_a_g_o: this is not an asterisk RTP problem |
| 16:31.13 | ManxPower | coppice, I agree. |
| 16:31.20 | ManxPower | m_a_g_o, what provider are you using? |
| 16:31.32 | tzanger | m_a_g_o: your far end that is not blanking out the audio when DTMF is being sent as RFC2833 is broken |
| 16:32.03 | ManxPower | m_a_g_o yes... that is clearly the problem, if dialing from a PSTN analog phone and setting it in MUTE while dialing the tones, it works just fine. |
| 16:32.03 | ManxPower | m_a_g_o not MUTE, duplicate tones |
| 16:32.54 | m_a_g_o | yes, I know but are all devices supposed to blank the audio when receiving RTP Event tones? |
| 16:33.25 | tzanger | m_a_g_o: no. You can have regular voice energy during RFC2833 DTMF, it's not up to the receiver to discern. |
| 16:34.06 | ManxPower | m_a_g_o, no all else fails, post a bounty to get someone to write a patch to do what you want. |
| 16:34.38 | m_a_g_o | so it would be solved if Asterisk was the one to blank the audi around an RTP Event... the RTP Event will arrive first than the audio, I believe... |
| 16:34.40 | ManxPower | I know the Packet8 devices (before Packet8 locked them down) had an option to mute or not mute audio when sending RFC2833 DTMF) |
| 16:35.05 | ManxPower | m_a_g_o, what provider do you use? |
| 16:35.10 | *** part/#asterisk ckruetze (n=ckruetze@i577A01D6.versanet.de) |
| 16:35.18 | m_a_g_o | new global telecom |
| 16:36.05 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 16:36.46 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 16:38.07 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 16:38.15 | m_a_g_o | I really wouldn't call them Evil... they just don't blank audio during DTMF tones... Asterisk doesn't either... |
| 16:39.14 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:39.47 | mog_work | kram! |
| 16:39.51 | kram | hi moggle |
| 16:40.01 | m_a_g_o | anyway, I have already prepaid minutes with them, so... isn't there any hope? |
| 16:40.04 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 16:40.04 | mog_work | how is the gplcon |
| 16:40.06 | Cresl1n | kram!!!! |
| 16:40.15 | ManxPower | m_a_g_o, no all else fails, post a bounty to get someone to write a patch to do what you want. |
| 16:40.17 | tzanger | m_a_g_o: yeah, use inband DTMF and not RFC2833. |
| 16:40.24 | kram | eh |
| 16:40.25 | kram | don't ask |
| 16:40.26 | kram | lol |
| 16:40.33 | hickins | Hello, |
| 16:40.33 | hickins | <PROTECTED> |
| 16:40.33 | hickins | <PROTECTED> |
| 16:40.33 | hickins | <PROTECTED> |
| 16:40.33 | hickins | iax.conf: |
| 16:40.33 | hickins | bindport=4569 |
| 16:40.33 | tzanger | gplcon? hah |
| 16:40.35 | hickins | bindaddr=0.0.0.0/0.0.0.0 |
| 16:40.37 | hickins | bandwidth=low |
| 16:40.39 | hickins | disallow=lcp10 |
| 16:40.39 | tzanger | hickins: don't do that |
| 16:40.41 | hickins | jitterbuffer=no |
| 16:40.43 | hickins | tos=lowdelay |
| 16:40.45 | *** kick/#asterisk [hickins!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted[asteria] (pastebin.ca) |
| 16:40.47 | m_a_g_o | thanks folks |
| 16:41.12 | mog_work | ouch twisted[asteria] |
| 16:41.20 | twisted[asteria] | mog_work, it's called flooding. |
| 16:41.24 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 16:41.33 | mog_work | yeah |
| 16:41.36 | mog_work | im with you |
| 16:41.51 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 16:42.02 | m_a_g_o | I'll se if I get dsp.c to drop some RTP Payload packets after an RTP Event, I'll let you know... |
| 16:42.10 | zoa | hey krambo |
| 16:42.10 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 16:42.12 | hickins | sorry guys, I got disconnected so I thought I lost what I've posted |
| 16:42.25 | kram | hey zoa, you got your ticket yet? |
| 16:42.42 | tzanger | m_a_g_o: if not I could probably do it |
| 16:42.50 | zoa | not yet |
| 16:43.11 | Damin | zoa: You going to be in S.F.? |
| 16:43.19 | zoa | no no |
| 16:43.32 | tzanger | hickins: regardless you don't post more than about 4 lines to the channel. Anything larger than that please use pastebin.ca |
| 16:43.50 | tzanger | man this voice dialer app for the treo is *COOL* |
| 16:43.59 | hickins | I see, appologies |
| 16:44.01 | jsharp | Which app? |
| 16:44.13 | tzanger | all the others you need to record the name... this one actually interprets your speech and matches it to the address book |
| 16:44.26 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 16:44.37 | jsharp | Niiiice. |
| 16:44.41 | tzanger | Treo Voice Dialer (rebranded voice dialer from VoiceSignal |
| 16:44.47 | eliel | Hello, I would like to know if there exists a sipura specific hardware channel to ask something... |
| 16:44.49 | tzanger | and it's pretty damn accurate too |
| 16:45.06 | tzanger | it'll do numeric matching too "Dial two nine one five one one two" |
| 16:45.14 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 16:45.21 | jsharp | Ohfudge. 650 only. |
| 16:45.33 | coppice | tzanger: the voice dialer in most cellphones works pretty well for names. |
| 16:45.59 | tzanger | coppice: all the voice dialers I've used before required me to record the names first |
| 16:46.09 | tzanger | coppice: i.e. I could say "shithead" and have it dial my brother |
| 16:46.28 | tzanger | with this one though if I say "shithead" it will try to match to the text "shithead" in my address book |
| 16:46.45 | Cresl1n | coppice: did you see anything obvious I might have broken in my up-ported version of your t.38 patch? |
| 16:47.17 | trixter | tzanger: which phone is that? all I have seen are the first you described |
| 16:47.24 | *** join/#asterisk Strom_C (n=strom@216-80-66-245.lem-bsr1.chi-lem.il.cable.rcn.com) |
| 16:47.31 | trixter | course in the last couple years embedded systems have seen a big jump in cpu power ... |
| 16:47.34 | tzanger | trixter: this is the treo voice dialer app by voicesignal for the treo650 |
| 16:47.57 | trixter | ahh ok |
| 16:48.03 | coppice | Cres11n: I didn't look |
| 16:48.09 | trixter | well the tro has a beefier cpu than a lot of cellphones I have seen ... |
| 16:48.17 | jsharp | Who'd have imagined a 300Mhz ARM processor in a phone. |
| 16:48.36 | trixter | I havea 400MHz pxa in my ipaq and use that like a phone from time to time |
| 16:48.46 | jsharp | Ohwait. Its not an ARM. Its the PXA270. |
| 16:48.47 | anderiv | hey all - I just installed a TE100P in my asterisk box. I'll be getting a T1 in next week. In the meantime, is there any testing I could do via a T1 loopback? |
| 16:48.52 | RoyK | zoa: ding? |
| 16:49.00 | tzanger | anderiv: what kind of testing are you interested in |
| 16:49.20 | *** part/#asterisk SYS64738 (n=giaco@host230-254.pool81123.interbusiness.it) |
| 16:49.36 | anderiv | tzanger: oh - just to make sure the card's up and running. Not sure if it's possible, but maybe send a call out through one channel and back in? |
| 16:50.14 | tzanger | plug in a loopback plug and see if it goes green |
| 16:50.18 | tzanger | (the LED, not the plug) |
| 16:50.35 | eliel | I'm having a problem with the sipura3000, when i make an outgoing call (using the pstn line) it answers my call before the call is completed (Voip answer delay), is there a way to 'delay' this answer after the callee really answers my call? |
| 16:51.00 | m_a_g_o | tzanger, I'm in dsp.c, exactly in the digit recognition section... I am not very familiar with the code, so I could use your advice =) |
| 16:51.15 | eliel | and asterisk thinks the call start befores it really starts |
| 16:51.27 | *** join/#asterisk TuRKisH (n=TimsaH@85.106.171.212) |
| 16:52.18 | jsharp | The sipura can't tell when the called party answers, so it considers the call to be "answered" as soon as it dials out. |
| 16:52.19 | m_a_g_o | I think that adding a conditional within thos Ifs asking for if DTMFMODE != inband would do it... |
| 16:52.24 | jsharp | There's no answer supervision. |
| 16:52.45 | tzanger | m_a_g_o: you're too low level |
| 16:52.57 | *** join/#asterisk javar_ (n=javar@69.79.217.237) |
| 16:52.57 | tzanger | m_a_g_o: first things first, are you using "t" or "T" dial options? |
| 16:53.34 | eliel | jsharp: so, is there a way with asterisk to detect human voice (backgrounddetect?) |
| 16:53.37 | m_a_g_o | I'm not sure what you're asking |
| 16:53.48 | m_a_g_o | as for timeout? |
| 16:53.50 | tzanger | m_a_g_o: check out bug 5970 first |
| 16:53.59 | m_a_g_o | ok... |
| 16:54.52 | RoyK | hm |
| 16:55.09 | RoyK | with asterisk 1.2 i get really bad dtmf reception |
| 16:56.27 | RoyK | users complain about having to dial dtmf SLOOOOOOWLY |
| 16:57.03 | tzanger | RoyK: odd |
| 16:57.15 | twisted[asteria] | RoyK, been using 1.2 here fine... even quick auto-redialers work great |
| 16:58.32 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:58.35 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:01.53 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 17:02.12 | RoyK | twisted[asteria]: when dialing from pstn i really need to dial dtmf slowly |
| 17:05.10 | ManxPower | RoyK, do you have relaxdtmf=yes set? |
| 17:05.20 | ManxPower | If so, don't do that. |
| 17:05.43 | ManxPower | If not, play around with your rxgain if on PRI or txgain and rxgain if on analog. |
| 17:06.26 | ManxPower | I had a problem with bad DTMF, turned out the gains were out of whack and causing distortion. |
| 17:07.39 | RoyK | yes |
| 17:07.49 | RoyK | i tried with and without that |
| 17:07.56 | RoyK | but it seems i need to Answer the call |
| 17:08.04 | anderiv | tzanger: well I plugged in a loopback cable, but no lights. A T1 loopback should be 1->4 and 2->5, right? |
| 17:08.04 | RoyK | Progress seems to be the problem |
| 17:08.24 | RoyK | and i don't really want to answer it...... charging money before dialling |
| 17:09.00 | tzanger | anderiv: correct. do you have the wct1xxp driver loaded? The card is "dead" until it's driver's loaded and ztcfg is run |
| 17:09.01 | jsharp | 1 to 5, 2 to 4 |
| 17:09.01 | Trazz | just talked to broadvoice and for every line you need from them you need to setup a seperate account and pay the 40$ activation fee. i need to find a carrier today that has similar priced service that wont bend me over on activation fees.. anyone? |
| 17:09.21 | ManxPower | RoyK, UM, you have to answer if you want to receive stuff from the caller |
| 17:09.22 | Trazz | so 5 lines is 200$ in fees plus the first month, etc.. |
| 17:09.28 | anderiv | tzanger: driver's loaded...hadn't run ztconfig yet. I'll get on that. |
| 17:09.40 | Math` | Trazz: they're fuckers at that price |
| 17:09.43 | anderiv | jsharp: 1->5, 2->4 huh? |
| 17:09.48 | jsharp | Yup. |
| 17:09.53 | ManxPower | Trazz, Do you REALLY want to trust your business to rely on the internet? |
| 17:09.55 | Trazz | yes i almost pulled the trigger until i found that |
| 17:10.20 | ManxPower | Trazz, I use Teliax. |
| 17:10.22 | Math` | Trazz: how many lines do you need? |
| 17:10.25 | RoyK | ManxPower: no, you don't |
| 17:10.30 | Math` | Trazz: and which are code are you in |
| 17:10.37 | RoyK | ManxPower: Progress works with dtmf in |
| 17:10.42 | ManxPower | But not for my clients. For my clients I put in PSTN at each location |
| 17:10.42 | RoyK | by standard |
| 17:10.44 | RoyK | it should at least |
| 17:10.45 | Trazz | i am in 312/773/630 and only need 1-3 lines |
| 17:11.10 | ManxPower | RoyK, Weird. |
| 17:11.19 | Trazz | ManxPower, i want to have phone service as convenience and wont be doing lots of calls |
| 17:11.24 | ManxPower | RoyK, which standard? |
| 17:11.45 | Math` | whats fun with standards is that there are plenty of them |
| 17:11.47 | Trazz | ManxPower, mainly customers use email and website today but we do get calls for inquiries |
| 17:11.48 | RoyK | eye ess dee enn? |
| 17:11.51 | ManxPower | Trazz, If you can put up with occasional downtime then use VoInternet. |
| 17:12.13 | Trazz | Math, do you have solution? |
| 17:12.17 | Trazz | i will check vointernet.. |
| 17:12.22 | Trazz | do you have site? |
| 17:12.25 | *** join/#asterisk Assid (n=assid@203.115.64.5) |
| 17:12.43 | ManxPower | RoyK, As I understand it telcos only allow one way audio before you answer. And since ISDN does not use OOB DTMF.... |
| 17:13.22 | ManxPower | Trazz, I refer to VoInternet (VoIP over the internet) to distinguish between that and VoLAN (VoIP over LAN) since you can control your lan, but you can't control the internet. |
| 17:13.37 | Trazz | ya i know.. |
| 17:13.48 | ManxPower | As I said, I use Teliax |
| 17:13.49 | Trazz | Math, any suggestions on who to go with ? |
| 17:13.50 | RoyK | ManxPower: er.. |
| 17:13.51 | *** join/#asterisk Defraz (i=t0tal@72.24.26.215) |
| 17:13.57 | RoyK | ManxPower: you may have a point there |
| 17:14.09 | RoyK | ManxPower: but it does indeed work with only Progress here |
| 17:14.09 | Trazz | i am chekcing out teliax now |
| 17:14.13 | ManxPower | RoyK, Of course DIALING on ISDN doesn't use DTMF at all, it uses setup messages. |
| 17:14.32 | RoyK | this is dial in, 'please enter card number.......' |
| 17:14.34 | RoyK | etc |
| 17:14.52 | RoyK | so Progress, Playback, Background etc |
| 17:14.57 | ManxPower | RoyK, *nod* outbound audio before answer. |
| 17:15.08 | RoyK | ISDN Early Media |
| 17:15.17 | SwK[Work] | early media is over rated |
| 17:15.26 | RoyK | of course it's not |
| 17:15.38 | ManxPower | That works for many carriers, but as far as I know most carriers don't allow inbound audio because if they did you could just not answer the call and so not pay for the call. |
| 17:15.41 | Cresl1n | RoyK: early media doesn't guarantee a two way audio path |
| 17:15.45 | anderiv | tzanger: for loopback testing, should timing be set to "1" for that span? |
| 17:16.03 | RoyK | Cresl1n: nope. that's not the point either |
| 17:16.04 | Cresl1n | RoyK: (Unless you've worked out some agreement with your telco) |
| 17:16.08 | Trazz | ManxPower any others you can recommend? |
| 17:16.34 | ManxPower | Trazz, Teliax is the only one that I use the has DIDs in many places. I also use NuFone, but not for DIDs |
| 17:18.03 | *** join/#asterisk roulduke_ (i=qqe2rbic@p508D1832.dip0.t-ipconnect.de) |
| 17:18.21 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:18.54 | Cresl1n | RoyK: so you're problem is that * has difficulty detecting DTMF before you Answer() the line? |
| 17:18.54 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 17:19.48 | zoa | woohoo |
| 17:20.01 | zoa | we patched asterisk to send audio every 20ms without receiving rtp :) |
| 17:20.22 | justinu | i have that patch too |
| 17:21.06 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 17:22.48 | eKo1 | send audio to who? |
| 17:23.55 | ManxPower | zoa, is that the "get timing from zaptel" patch? |
| 17:23.56 | justinu | i use it for early media on sip |
| 17:24.33 | *** join/#asterisk BuddaBoy (n=irc@62.69.89.38) |
| 17:25.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:25.44 | rhousand | is it possable to transfer a cal to a hard line phone number? |
| 17:26.33 | darkskiez | rhousand: yes |
| 17:27.58 | rhousand | Darkskiez: I am new to * and do not wount to wast your time.. do you know a link that i cauld find some information on doing this? |
| 17:28.20 | darkskiez | rhousand: there is no such information, as it is identical to transferring to another extension |
| 17:28.56 | zoa | ManxPower: dunno, gimme the mantis number please |
| 17:29.10 | jsharp | Yes. * makes no distinction between "Extensions" and "hard line phone numbers". Everything is just an extension. |
| 17:29.14 | ManxPower | zoa, no idea. |
| 17:29.21 | rhousand | greate thanks |
| 17:29.42 | zoa | its something to fix the problems when connecting to a cirpak gateway |
| 17:30.30 | tzanger | anderiv: won't matter |
| 17:30.38 | ManxPower | zoa, it would also fix the problem of outgoing audio being jittery because the incoming audio is jittery |
| 17:30.39 | darkskiez | everytime i think i've totally sussed out * and nat, it bites me in the ass again |
| 17:31.03 | jsharp | That's cause NAT sucks. |
| 17:31.14 | fugitivo | i like nat |
| 17:31.24 | ManxPower | I just almost never have NAT problems |
| 17:31.35 | fugitivo | the problem is not NAT, it's sip and asterisk |
| 17:31.37 | zoa | ManxPower: no not really it will only give you ringing now |
| 17:31.43 | Trazz | Manxpower, looks like nufone is michigan only ... i need something for chicago #'s |
| 17:31.44 | zoa | that would be the real patch |
| 17:31.57 | zoa | we made a quick patch (36 hours) to fix this thing that bothered us |
| 17:32.11 | darkskiez | rite, home |
| 17:32.11 | zoa | but your problem would be fixed with the jitter buffer |
| 17:32.28 | ManxPower | Trazz, That's why I don't use Nufone for non-mich DIDs. |
| 17:32.46 | ManxPower | Trazz, try out Teliax. Get an account with a few $'s on it and try them out. |
| 17:33.10 | ManxPower | You would not get married without first meeting the other person, why bet your company on an ITSP that you've never used. |
| 17:33.31 | ManxPower | ALL ITSPS SUCK. Teliax just seems to just less than most others. |
| 17:33.55 | fugitivo | world sucks |
| 17:34.17 | jsharp | Gravity doesn't exist. The Earth just sucks. |
| 17:34.20 | *** join/#asterisk FastJack (i=fastjack@p5091CAF2.dip.t-dialin.net) |
| 17:34.31 | *** join/#asterisk ToTo (n=ToTo@host71-159.pool874.interbusiness.it) |
| 17:35.24 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 17:38.26 | *** join/#asterisk secure75 (n=mic@217.237.169.66) |
| 17:42.08 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 17:45.52 | jbalcomb | is it /normal/ for Asterisk to segment a few minutes after a reload or two? |
| 17:45.52 | ManxPower | jbalcomb, no |
| 17:45.53 | ManxPower | define "segment"? |
| 17:45.57 | ManxPower | do you mean Seg Fault? |
| 17:46.02 | ManxPower | And what version of Asterisk? |
| 17:47.01 | jbalcomb | ManxPower yeah, seg fault; asterisk 1.2.1 |
| 17:47.46 | jbalcomb | ManxPower wondering if its related to -vvvvv being on, the changes themselves, or the vast amount of WARNING and NOTICE msgs im getting. |
| 17:47.52 | tzanger | jbalcomb: what distro |
| 17:47.58 | tzanger | and were did you get the 1.2.1 asterisk |
| 17:48.01 | *** join/#asterisk pnlarsson (n=e4599069@c83-248-7-150.bredband.comhem.se) |
| 17:48.15 | justinu | jbalcomb: what kinda warnings? |
| 17:48.44 | bkw_ | doubtful the distro has anything to do with asterisk sucking |
| 17:48.56 | tzanger | bkw_: not if he's riced it out with default Gentoo settings |
| 17:49.22 | fugitivo | gentoo is stable |
| 17:49.28 | *** join/#asterisk c3r3al-kill3r (n=Nbm@87-196-45-240.net.novis.pt) |
| 17:49.29 | tzanger | fugitivo: it is if you're not an idiot |
| 17:49.33 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:49.40 | fugitivo | any distro is unstable if you're an idiot |
| 17:49.49 | tzanger | but gentoo is the #1 distro for attracting pizza-faced 13 year olds who "tweak out their shit" |
| 17:50.14 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:50.31 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 17:51.23 | [TK]D-Fender | jbalcomb : Your company's * setup just bombed? |
| 17:51.55 | jbalcomb | tzanger im being told we downloaded the asterisk from digiums site |
| 17:51.59 | justinu | lol @ riced it out |
| 17:52.18 | justinu | jbalcomb: how about pasting some of your warnings/errors to pastebin.com? |
| 17:52.28 | tzanger | jbalcomb: ok. what kind of system is it on (cpu, etc.) and distro. It's not common for * to crash at all |
| 17:52.48 | jbalcomb | Whitebox |
| 17:52.48 | jbalcomb | CPU: Dual PIV Xeon 3.0 Ghz |
| 17:52.48 | jbalcomb | MEM: 2 GB |
| 17:52.48 | jbalcomb | HDD: 80 GB SCSI |
| 17:52.48 | jbalcomb | NIC: 1 Gbps (2) |
| 17:52.48 | jbalcomb | OS: Debian 1:3.3.5-12 (Linux version 2.6.8-2-686-smp) |
| 17:53.07 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:53.12 | tzanger | jbalcomb: ok. did you have an issues building asterisk? compiler errors or anything? |
| 17:54.23 | tzanger | are any other apps or daemons on this box crashing? Is it repeatable? yes it's a lot of questioning... but these are important things to know |
| 17:55.17 | jbalcomb | tzanger it seems that only Asterisk is crashing. it has happened twice today. not sure if i want to try and repeat it. (anxious boss) |
| 17:55.52 | jbalcomb | justinu im pulling the errors now.. |
| 17:56.22 | SwK[Work] | was there a AMI bug that would cause manager disconnects after issuing a originate? |
| 17:56.32 | DrWho17 | tzanger: gentoo is out, ubuntu is cool these days |
| 17:56.41 | jbalcomb | justinu http://pastebin.com/510149 |
| 17:56.54 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
| 17:57.17 | jbalcomb | tzanger i was not here when the box was built so I am not sure about that |
| 17:57.25 | justinu | jbalcom: it looks like you have issues in sip.conf |
| 17:57.32 | DrWho17 | unload all necessary modules and work back from their |
| 17:57.32 | justinu | and possible a bunch of other config file issues |
| 17:57.46 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc12c.dialup.mindspring.com) |
| 17:57.48 | DrWho17 | well unecessary, heh |
| 17:58.03 | jbalcomb | DrWho17 ok, i was looking at that this morning, found the show modules command. lots of mods showing 0 use. |
| 17:58.30 | jbalcomb | DrWho17 im atleast going to unload the ones that asterisk notes on startup that it wont use |
| 17:58.55 | DrWho17 | sure, maybe look at the debug as well, see what happens before it goes poof |
| 17:59.16 | jbalcomb | justinu yeah, [tk]defender gave me the heads up on one issue of some deprecated DBGet/DBPut code. |
| 17:59.32 | DrWho17 | my experience with asterisk, is that it is pretty solid |
| 17:59.35 | jbalcomb | DrWho17 sadly, i am not sure how to do that or what im looking for. |
| 17:59.44 | DrWho17 | in /var/log/asterisk |
| 17:59.55 | justinu | jbalcomb: this looks more fundamental than that. |
| 17:59.58 | DrWho17 | should be some nice logs to poke through |
| 18:00.11 | justinu | jbalcomb: like why is there a blank host in sip.conf? |
| 18:01.01 | DrWho17 | you you are storing asterisk internal database externally as well? |
| 18:01.46 | jbalcomb | justinu what indicates a blank host and is there a way to find it quickly in the 1666 lines of sip.conf? |
| 18:02.01 | jbalcomb | DrWho17 you mean in MySQL or something of the sort? if so, no. |
| 18:02.27 | DrWho17 | ok, that would be odbcdbget/dbput |
| 18:02.28 | jbalcomb | DrWho17 ok, yeah, im on with the logs. i thought you meant doing something with the core dump file. |
| 18:03.09 | [TK]D-Fender | DrWho17 : No he's just using the internal DB1 for some dial-plan controls. |
| 18:03.30 | jbalcomb | [TK]D-Fender i updated that code but i still see the message |
| 18:03.39 | DrWho17 | sure, that path isn't his problem |
| 18:03.45 | jbalcomb | Jan 17 12:29:28 WARNING[20089]: ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 18:03.45 | jbalcomb | =0800 |
| 18:03.45 | jbalcomb | ^ |
| 18:03.45 | jbalcomb | Jan 17 12:29:28 WARNING[20089]: ast_expr2.fl:180 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. |
| 18:03.45 | jbalcomb | <PROTECTED> |
| 18:03.46 | [TK]D-Fender | jbalcomb : What code? |
| 18:04.00 | justinu | jbalcomb: search for host= |
| 18:04.38 | *** join/#asterisk [1]EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 18:06.57 | jbalcomb | justinu no blank host= entry. perhaps a section that doesnt have a host statement? |
| 18:07.20 | jbalcomb | [TK]D-Fender exten => s,4,GotoIf($[${DB(dnd/SIP/63)}=0800]?8) |
| 18:07.32 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 18:07.39 | *** join/#asterisk zamsler_ (n=zamsler@c-67-184-234-6.hsd1.il.comcast.net) |
| 18:07.40 | jbalcomb | [TK]D-Fender OLD CODE;exten => s,5,GotoIf($[${dnd} = 0800]?9:6) |
| 18:08.11 | [TK]D-Fender | jbalcomb that "current one is bad |
| 18:08.24 | justinu | jbalcomb: not sure, but those errors look pretty weird |
| 18:08.33 | [TK]D-Fender | jbalcomb : PM |
| 18:09.56 | rob0 | [TK]D-Fender: you still have an unmatched \" |
| 18:10.29 | ManxPower | jbalcomb, sounds like you have an error in your expressions. |
| 18:11.00 | ManxPower | jbalcomb, you forgot a space around the = |
| 18:11.07 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 18:11.10 | jbalcomb | ManxPower yeah, that code is wierd because it was written to match the dnd on/off function of IPSwitchboard |
| 18:11.51 | jbalcomb | ManxPower i looked in the README.variables but i didn't 'get' it so much |
| 18:12.13 | [TK]D-Fender | jbalcomb : Your fix isn't right and doesn't match what I suggested yesterday. |
| 18:12.20 | ManxPower | always put a space around the = |
| 18:12.35 | jbalcomb | ManxPower hrmm.. the wiki has Set(DB(family/key)=${foo}) |
| 18:12.52 | ManxPower | jbalcomb, SET is different from GotoIf. |
| 18:12.57 | *** join/#asterisk _nvrs (i=RUR@HSE-Montreal-ppp3469494.sympatico.ca) |
| 18:13.03 | [TK]D-Fender | jbalcomb : I just sent you a private message |
| 18:13.07 | ManxPower | SET is an assignment, Gotoif is a comparison |
| 18:13.29 | DrWho17 | been a long time since I've compiled up an asterisk, if I want to use the addons, is it just a matter of running make in the addons dir. provided ../asterisk is in the correct path |
| 18:13.31 | ManxPower | I should have said "Always put a space around = when doing a comparison" |
| 18:13.41 | jbalcomb | ManxPower ok, i gotcha. looking at the wiki now and i see what you're saying. |
| 18:13.42 | DrWho17 | and of course all my mysql includes are installed |
| 18:13.51 | DrWho17 | no more manual patching |
| 18:14.05 | [TK]D-Fender | jbalcomb : Pastebin the whole thing up again, we need to make sure both sides match |
| 18:14.09 | ManxPower | jbalcomb, want an example of some complex dialplan stuff with gotos, comparisons, and sets? |
| 18:14.47 | jbalcomb | ManxPower I think thats what I have! ;) Actually it would be nice to see something done right instead of this hackjob I have. |
| 18:14.56 | [TK]D-Fender | ManxPower : I gave him a corrected sample yesterday but the patch job seems incomplete. his setup was decent though depricated for 1.2.1 |
| 18:14.59 | jbalcomb | [TK]D-Fender I PMd ya back |
| 18:15.33 | [TK]D-Fender | jbalcomb : Didn't work out right. Pastebin your extensions.conf section for all that again and I'll redo the fix for you |
| 18:15.46 | jbalcomb | [TK]D-Fender ok |
| 18:16.39 | *** join/#asterisk nicksin (n=nicl@gateway.cpcplus.co.za) |
| 18:16.49 | nicksin | Good evening all. |
| 18:16.57 | nicksin | Anyone here from South Africa ? |
| 18:17.26 | ManxPower | My standard exten macro example: http://pastebin.ca/37098 |
| 18:17.40 | ManxPower | If you read it your brain may explode. Don't blame me if it does. |
| 18:17.50 | [TK]D-Fender | ManxPower : His setup was clean if repetitive and only needed a quick fix |
| 18:18.00 | ManxPower | jbalcomb, um, EVERYTHING in extensions.conf is a hackjob 8-) |
| 18:18.13 | jbalcomb | [TK]D-Fender http://pastebin.com/510179 is my applications.conf which is included in extensions.conf |
| 18:18.38 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 18:18.52 | rob0 | <== boom (that was my head) |
| 18:18.55 | [TK]D-Fender | BAD -> exten => _*60,n,Set(DB(dnd/SIP/60)=0800) |
| 18:19.04 | [TK]D-Fender | GOOD -> exten => _*60,n,Set(DB(dnd/60)=0800) |
| 18:19.16 | nicksin | I need some help in Getting a Duxbury card working with an old style PABX |
| 18:19.34 | jbalcomb | [TK]D-Fender http://pastebin.com/510182 is my acd_queues.conf also included in my extensions.conf |
| 18:19.38 | [TK]D-Fender | jbalcomb : now I need the quque section for the gotoif |
| 18:19.58 | Trazz | TK, it looks like my * is answering the call and playing the greeting while the voip line is still ringing and once answered its not taking my dtmf.. |
| 18:20.07 | *** join/#asterisk gaspiz (i=gaspi@86.34.6.164) |
| 18:20.08 | [TK]D-Fender | jbalcomb : BAD -> exten => s,6,GotoIf($[${dnd} = 0800]?10:7) |
| 18:20.38 | [TK]D-Fender | jbalcomb: GOOD -> exten => s,6,GotoIf($[${DB(dnd/60)}=0800]?10) |
| 18:20.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:20.53 | [TK]D-Fender | jbalcomb : And then remove the useless DBGET in there and renumber things |
| 18:21.45 | [TK]D-Fender | and change this - exten => _**60,n,DBdel(dnd/SIP/60) |
| 18:22.10 | jbalcomb | [TK]D-Fender ok, i guess i should have asked yesterday but that dnd/SIP/60 is the format used by IPSwitchboard whos function I have to match up with. Does that change the GOOD/BAD/NECESSARY scenerio? |
| 18:22.11 | [TK]D-Fender | to : exxten => _**60,n,Set(DB(dnd/60)=none) |
| 18:22.30 | *** part/#asterisk secure75 (n=mic@217.237.169.66) |
| 18:22.45 | *** join/#asterisk SuidBit (n=LinuxSec@darwin.fundanet.br) |
| 18:22.58 | jbalcomb | [TK]D-Fender yeah, see thats what I would think is the thing to do but I'm screwed on the issue for now because we are using IPSwitchboard |
| 18:23.04 | [TK]D-Fender | jbalcomb : the ipswitchboard side you are doing just fakes status, and has nothing to do with the ASTDB stuff you are setting up to prevent the queu from being called |
| 18:23.16 | jbalcomb | [TK]D-Fender they have to be able to set and unset DnD from the phone or IPSwitchboard |
| 18:23.22 | [TK]D-Fender | jbalcomb : two very seperate pieces of the same puzzle. |
| 18:23.48 | [TK]D-Fender | jbalcomb :hold on |
| 18:24.11 | jbalcomb | [TK]D-Fender not sure i follow you. when they set DnD in IPSwitchboard is creates that dnd/SIP/60 entry in the Asterisk DB |
| 18:24.40 | jbalcomb | [TK]D-Fender if they so choose they can hit **60 to turn DnD off instead of using IPSwitchboard |
| 18:24.58 | [TK]D-Fender | :/ |
| 18:25.27 | [TK]D-Fender | jbalcomb : Ok wierd, not sure if you can do that now though since you have an extra slash in there |
| 18:25.52 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 18:26.20 | jbalcomb | [TK]D-Fender yeah, its definitely wierd. who would use the damn field deliminator in thier key name.. |
| 18:26.58 | [TK]D-Fender | jbalcomb : You need another interface to this... would be an easy PHP scripted web page to make.... |
| 18:27.20 | jbalcomb | [TK]D-Fender for now it works though and im just trying to get all my duct tape in order before i start restructuring anything |
| 18:27.34 | [TK]D-Fender | jbalcomb : Not sure what to suggest to patch this.... |
| 18:27.44 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:28.01 | [TK]D-Fender | your original form looks as close to right as it should, but I'm not sure if the DB function likes that extra slash now.. |
| 18:28.33 | ManxPower | one would assume that if IPSwitchboard is actually working, then it would be OK. |
| 18:28.54 | jbalcomb | [TK]D-Fender yeah, I'm gonna put those spaces in there an see if that helps |
| 18:29.32 | *** join/#asterisk zamsler__ (n=zamsler@c-67-184-234-6.hsd1.il.comcast.net) |
| 18:29.42 | RoyK | -.-. --.- |
| 18:29.44 | jbalcomb | I'm just hoping at this point if I can eliminate the 10 WARNING/NOTICE / second I'll be able to better focus on figuring out everything else. |
| 18:29.47 | [TK]D-Fender | jbalcomb : better yet : make in exten to retreive it and just NoOp it to consaole |
| 18:31.00 | ManxPower | I still say you need a space around = in a gotoif |
| 18:31.28 | jbalcomb | ManxPower I'm putting it in and reloading now. |
| 18:32.45 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 18:33.25 | [TK]D-Fender | ManxPower : I never did... |
| 18:34.34 | [TK]D-Fender | ManxPower : just a sample from home - http://pastebin.ca/37100 |
| 18:35.02 | [TK]D-Fender | ManxPower : It and the rest are just fine. |
| 18:35.39 | ManxPower | I just tried it and it failed with the same error |
| 18:35.46 | ManxPower | [TK]D-Fender jbalcomb: GOOD -> exten => s,6,GotoIf($[${DB(dnd/60)}=0800]?10) |
| 18:36.57 | ManxPower | ,GotoIf($["${DB(dnd/60)}"="0800"]?10) for when the value is empty or does not exist |
| 18:37.32 | gaspiz | i'm playing back a file using the background command from a php agi script. If during the playback the caller hangs up my agi becomes a zombie process |
| 18:37.42 | gaspiz | can anyone help? |
| 18:37.51 | ManxPower | gaspiz, that's because you are not traping SIG INT or whatever it is. |
| 18:38.00 | ManxPower | gaspiz, this was talked about on the mailing lists. |
| 18:38.11 | gaspiz | i'll read |
| 18:38.14 | gaspiz | thanks |
| 18:39.02 | *** join/#asterisk cyber (n=kani@220.247.248.242) |
| 18:40.28 | ManxPower | ~mailinglist |
| 18:40.30 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 18:41.36 | jbalcomb | ManxPower I would want the """"s then because I'm checking for a DB entry that most likely doesn't exist due to its being DBDel'd when DnD is set off |
| 18:41.48 | Zodiacal | im trying to setup asterisk for the first time, i disconnected a few lines from my old pbx system and when i pluged them back in, my ring down (hunt?) doesn't work. is that verizon's fault or my old pbx's? |
| 18:41.50 | *** join/#asterisk FastJack (i=fastjack@p5091CAF2.dip.t-dialin.net) |
| 18:41.59 | ManxPower | jbalcomb, did the space around = help or not? |
| 18:42.07 | jbalcomb | ManxPower doesn't seem to have |
| 18:42.27 | ManxPower | jbalcomb, try quotes then. Maybe 1.2 removed the requirement for spaces around =. |
| 18:42.34 | ManxPower | I'm sure they don't hurt, however. |
| 18:42.50 | jbalcomb | ManxPower http://pastebin.com/510226 is my call as I see it in the CLI at verbose 99 |
| 18:43.45 | Trazz | ManxPower, i dial in from outside and it never sends to voice mail when i hit extension.. |
| 18:43.49 | [TK]D-Fender | ManxPower : I guess you need something on the left & right of the = at all times for that not to bomb? |
| 18:44.48 | ManxPower | [TK]D-Fender, ya think? |
| 18:44.53 | *** join/#asterisk sac|h0p|werk (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
| 18:44.55 | jbalcomb | ManxPower [TK]D-Fender the 1st example in the wiki has quotes AND spaces |
| 18:45.21 | Trazz | jbalcomb, what is the page with the example? |
| 18:45.22 | Zodiacal | any ideas? |
| 18:45.32 | jbalcomb | Trazz the page for GotoIf command |
| 18:45.37 | Trazz | ok |
| 18:45.38 | [TK]D-Fender | jbalcomb : Try them all, but before even bothering with the gotoif, do the noOp of jsut the DB() that pulls the value first |
| 18:46.12 | *** join/#asterisk Charon212 (n=nschmidt@printer.fxserver.com) |
| 18:46.18 | [TK]D-Fender | ManxPower : Dunno on the requirement of left and right sides. It could be valid depending on implementation. |
| 18:46.20 | Zodiacal | verizon controls the hunt right? |
| 18:46.23 | Trazz | how hard is it to have the ivr ask for a ticket # and then have it go to a mysql query and say the status? |
| 18:46.30 | jbalcomb | [TK]D-Fender oh, thats just for purposeful debugging? |
| 18:46.49 | jbalcomb | Trazz doesnt seem too hard at all. |
| 18:46.50 | Charon212 | Hey all. Quick Question. I am setting up my IVR and I want one of the options to forward to an outside number through SIP. What is the string I would use? |
| 18:47.04 | infinity1 | is it recommended to always use qualify when a fone is behind nat? |
| 18:47.25 | [TK]D-Fender | jbalcomb : yes, rather helpful. Make sure you can ever retreive the value in the first place |
| 18:47.35 | ManxPower | infinity1, yes |
| 18:47.53 | [TK]D-Fender | infinity1 : yes otherwise most nat routers will forget where to send the UDP to |
| 18:47.55 | jbalcomb | Trazz just grab input, save variable, query mysql, use entry in database to match preset audio file, play |
| 18:48.21 | [TK]D-Fender | Charon212 : depends on how you set up your SIP destination point. |
| 18:48.27 | Trazz | jbalcomb, any examples you can point me to ? |
| 18:48.30 | *** join/#asterisk NoVaZuR (n=novazur@ALamentin-104-1-13-110.w81-248.abo.wanadoo.fr) |
| 18:48.34 | NoVaZuR | hi |
| 18:48.35 | jbalcomb | [TK]D-Fender well, most of the time that value isn't going to be there because the DB family/key is deleted for DnD off status |
| 18:48.52 | jbalcomb | Trazz i am afraid not, all I have is theory. |
| 18:48.55 | ManxPower | Charon212, your question makes about as much sense as asking "How do I get from Birmingham to Atlanta?" |
| 18:48.58 | Trazz | heheheeh ok |
| 18:49.31 | ManxPower | The answer depends on how much money you have, if you have a car or not, maybe you are marathon runner and want to run the distance.... |
| 18:49.48 | gaspiz | ManxPower: about the zombie process generated if hangup comes while i'm executing a command. I didn't find any usefull info. can you explain briefly? |
| 18:50.13 | ManxPower | gaspiz, your AGI is ignoring the signal sent to it when the call hangs up. |
| 18:50.14 | [TK]D-Fender | jbalcomb : MAKE a value for it to find... |
| 18:50.33 | NoVaZuR | how is it possible to send a fax with OOo (for example) throw asterisk please ? I already have hylafax installed, and working with my last modem (replaced by x100p card). But I don't know how to tell it to use asterisk. |
| 18:51.19 | [TK]D-Fender | NoVaZuR : You'd have to have it output a TIFF or something and use the TxFAX command. |
| 18:51.21 | gaspiz | MAnxPower: thanx, will search about the chaching of signals, if I cant find anything will come back :P |
| 18:51.32 | [TK]D-Fender | NoVaZuR : Go start WIKI-ing |
| 18:51.50 | *** part/#asterisk Davey|Work (n=davey@unaffiliated/davey) |
| 18:51.55 | *** join/#asterisk r_evolution (i=_evoluti@12.155.106.12) |
| 18:52.01 | NoVaZuR | [TK]D-Fender: impossible to send directly from OOo so ? |
| 18:52.24 | NoVaZuR | [TK]D-Fender: I've already read all of that, but I don't understand very well |
| 18:53.22 | jbalcomb | Bored? Check on my CLI messages for the Reload!! http://pastebin.com/510245 |
| 18:53.22 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
| 18:53.33 | [TK]D-Fender | NoVaZuR : There is no magic "go" button.... * isn't a fax server |
| 18:53.48 | jbalcomb | FoIP |
| 18:53.59 | NoVaZuR | [TK]D-Fender: yes, but I thought hylafax could discuss with it :-( |
| 18:53.59 | ManxPower | gaspiz, see if the phpagi people have a mailinglist |
| 18:54.12 | r_evolution | im starting to think D-Fender lives in here O_o |
| 18:54.21 | [TK]D-Fender | jbalcomb : Looks like you might have double-INCLUDED a file in your dial-plan |
| 18:54.37 | [TK]D-Fender | NoVaZuR : Hylafx has nothing to do with *. |
| 18:54.57 | NoVaZuR | [TK]D-Fender: no, of course |
| 18:54.58 | [TK]D-Fender | r_evolution : No, but I die a little here each day ;) |
| 18:55.43 | ManxPower | gaspiz, perl, but same idea: http://lists.digium.com/pipermail/asterisk-users/2004-February/037481.html http://lists.digium.com/pipermail/asterisk-users/2004-February/037443.html |
| 18:55.52 | r_evolution | i die a little at work every day |
| 18:56.08 | dogtanian | lol |
| 18:56.11 | dogtanian | i know the feeling |
| 18:56.14 | *** join/#asterisk high| (n=mtryfoss@80.239.93.22) |
| 18:56.18 | jbalcomb | [TK]D-Fender im not seeing that. grep for include shows no doouble include. perhaps when the extensions.conf was split up code was duplicated. I'll loook into it. |
| 18:56.55 | dogtanian | r_evolution]: it's like having your soul chipped-away at with a chisel |
| 18:57.00 | *** join/#asterisk gaspiz (i=gaspi@86.34.6.164) |
| 18:57.09 | r_evolution | a chisel? |
| 18:57.14 | r_evolution | i was thinking more like |
| 18:57.18 | r_evolution | pointless questions |
| 18:57.29 | ManxPower | Quit and become a consultant! |
| 18:57.43 | r_evolution | "Hi... I know you have this black screen with white letters up in front of you... and i can see a BUNCH of those white letters you call 'code' on it... |
| 18:57.44 | r_evolution | but... |
| 18:57.46 | r_evolution | are you busy?" |
| 18:57.52 | wunderkin | r_evolution, are you on the phone? |
| 18:57.55 | jbalcomb | shit... i just sent all calls for that customer directly to voicemail |
| 18:58.02 | NoVaZuR | [TK]D-Fender: thanks for all. Bye |
| 18:58.11 | r_evolution | only every now and again... and thats to get something |
| 18:58.12 | r_evolution | EXCEPT |
| 18:58.17 | *** part/#asterisk ruud_org (n=ruud@83.96.138.21) |
| 18:58.20 | r_evolution | for the uhm voip server we rolled out :-\ |
| 18:58.22 | gaspiz | ManxPower: do you know what signal sends the asterisk to the agi (php) on hangup? |
| 18:58.30 | *** join/#asterisk IRS (n=Nbm@87-196-10-189.net.novis.pt) |
| 18:58.39 | ManxPower | gaspiz, perl, but same idea: http://lists.digium.com/pipermail/asterisk-users/2004-February/037481.html http://lists.digium.com/pipermail/asterisk-users/2004-February/037443.html |
| 18:58.42 | *** part/#asterisk NoVaZuR (n=novazur@ALamentin-104-1-13-110.w81-248.abo.wanadoo.fr) |
| 18:58.51 | rob0 | Consultant: If you're not part of the solution, there is much money to be made in prolonging the problem. |
| 18:59.13 | ManxPower | rob0, Exactly! |
| 18:59.18 | *** join/#asterisk effape (n=nick@81.5.150.54) |
| 18:59.37 | r_evolution | haha |
| 18:59.44 | *** join/#asterisk nvrs (i=RUR@Quebec-HSE-ppp3614140.sympatico.ca) |
| 18:59.46 | r_evolution | rob0 seems to always chime in with some bit of good humor. |
| 19:00.54 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 19:01.50 | *** join/#asterisk silentfury_ (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 19:02.01 | silentfury_ | has anyone setup an audiocodes mp108 fxo gateway before? |
| 19:02.02 | darkskiez | asterisk is putting localhost in the connection information field in the invite SDP, what the hell am i doing wrong? |
| 19:02.17 | effape | me again :) - |
| 19:02.18 | Trazz | when i dial from outside and goo through my menu it rings my extension but never sends it to voice mail like it does on the inside |
| 19:02.31 | effape | Don't suppose anyone has any ideas how to stop this from happening - http://pastebin.com/510259 |
| 19:02.49 | znoG | OT question: how do you execute a command in vim and make the output go into vim? |
| 19:03.04 | ManxPower | darkskiez, stop using bindaddr. |
| 19:03.08 | jbalcomb | iCEBrkr i just came to understand what the X you added is doing.. |
| 19:03.19 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:03.55 | ManxPower | a2) you have a nat problem and a3) the IP addresses of your host are not resolvable (/etc/hosts or DNS) |
| 19:04.02 | nextime | anyone using chan_woomera for h323? |
| 19:04.04 | [TK]D-Fender | jbalcomb : thats to prevent the "blank" possibility |
| 19:04.55 | ManxPower | effape, what country are you in? |
| 19:04.57 | jbalcomb | [TK]D-Fender yeah, it just clicked that i'm evaluating that 0800 against that pull from the DB but if the family/key:value doesn't exist, its null. |
| 19:05.07 | jbalcomb | [TK]D-Fender atleast I'd think it would be null |
| 19:05.11 | effape | uk. It disconnects when i pick up the sip (xlite) |
| 19:05.35 | ManxPower | effape, sounds like you have analog DID lines or something like that. |
| 19:05.47 | [TK]D-Fender | jbalcomb : I was working it as though you were intending to MONITOR it from ipswitchboard, not use IPS to tollge it live like a fake extension |
| 19:06.00 | effape | yeah analog into tdm400p |
| 19:06.01 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 19:06.04 | ManxPower | effape, Asterisk is expecting the polarity reversal to signal hangup, not to signal answer. |
| 19:06.27 | ManxPower | effape, Um, Asterisk analog cards don't support wink or E&M, etc. |
| 19:06.48 | effape | i'm pretty new to all this so i'm just trying to get it all working. |
| 19:06.59 | effape | What should i do to sort it? |
| 19:07.19 | darkskiez | ManxPower: was the hosts file, cheeeers :) |
| 19:07.48 | ManxPower | effape, I have no idea. What happens if you plug an analog phone into the line and make a call? |
| 19:07.52 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 19:08.07 | effape | the line works fine. |
| 19:08.17 | r_evolution | could you not put hanguponpolaryswitch=no in the zapata.conf? |
| 19:08.19 | jbalcomb | [TK]D-Fender yeah, as quick as I can im gonna change that. dont know why they'd pick a value as confusing as 0800 to denote DnD status. It means 8 AM they told me.. |
| 19:08.44 | effape | yeah i did that and it doesn't hangup anymore - including when you put the sip phone down :( |
| 19:08.50 | ManxPower | effape, your extensive search of the mailing list archives and the Wiki did not turn anything up? |
| 19:08.59 | effape | it works upto that point though - with that option as no |
| 19:09.18 | effape | No. i've been banging my head against a wall for a week :) |
| 19:09.36 | effape | i had cidstart-polarity which resulted in no pickup |
| 19:09.36 | ManxPower | effape, ask on the mailing lists then. |
| 19:09.47 | r_evolution | eff... perhaps answeronpolarityswitch? |
| 19:09.58 | r_evolution | rather than hangup... ? |
| 19:10.17 | jbalcomb | [TK]D-Fender -- Executing GotoIf("Zap/27-1", "X0?8") in new stack |
| 19:10.40 | jbalcomb | [TK]D-Fender what is with the 0?8 |
| 19:10.41 | r_evolution | dont take my advice too seriously all the time :p I'm rather new at this myself... but I spend entirely too much time reading... |
| 19:10.46 | Charon212 | ok, I will revise my question a bit. I appologize. I am a bit new at this whole asterisk thing. |
| 19:10.49 | Charon212 | its probably a very silly problem but I cant dial out over my sip provider. I want to specify an option off my IVR to call a cell phone. What do I need to specify in my extentions.conf to call out over my sip provider. |
| 19:10.52 | [TK]D-Fender | jbalcomb : Forget the GotoIf and test for the VALUE. Just NoOpit.... |
| 19:10.58 | ManxPower | jbalcomb, that is what the expression inside the $[] evaluates to |
| 19:11.11 | Charon212 | so "Press 1 to reach... whatever" |
| 19:11.22 | r_evolution | Charon... can you dial out over your SIP provider not from inside the IVR? |
| 19:11.29 | ManxPower | Charon212, your questons are so basic nobody will answer them. |
| 19:11.31 | r_evolution | i.e. can you pick up a phone and dial? |
| 19:11.32 | ManxPower | ~docs |
| 19:11.34 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 19:11.45 | Charon212 | Yes I can dial out over the provider |
| 19:11.48 | r_evolution | Nah Manx... everyone needs help... he just needs to learn to read the wiki |
| 19:11.53 | r_evolution | then use the dial command in your dial plan :) |
| 19:12.04 | effape | sorry just on the phone brb |
| 19:12.24 | Charon212 | Ive looked at the wiki for the dial command and for the SIP syntax. Nothing tells me how to dial a number. Just SIP peers and IP phones |
| 19:12.37 | r_evolution | sure it does. |
| 19:12.44 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 19:12.59 | riddlebox | has anyone done any AGI scripting with python? |
| 19:13.17 | tainted_ | anyone have a 212 DID? :D |
| 19:13.40 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 19:13.49 | tainted_ | 212 DID? anyone? |
| 19:13.53 | silentfury | anyone ever set up an audiocodes mp-108 gateway? |
| 19:13.55 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 19:14.08 | jbalcomb | [TK]D-Fender like exten => s,4,NoOp($[${DB(dnd/SIP/63/)}]) |
| 19:14.40 | *** part/#asterisk SuidBit (n=LinuxSec@darwin.fundanet.br) |
| 19:15.25 | [TK]D-Fender | jbalcomb : like - exten => s,4,NoOp(${DB(dnd/SIP/63)}) |
| 19:17.03 | jbalcomb | [TK]D-Fender ok, damnit, got it. testing yet again and hopeing this stuff will start making sense soon. |
| 19:18.35 | brad_mssw | tainted_: err, junctionnetworks does |
| 19:19.17 | brad_mssw | (so does teliax, but they've been having issues) |
| 19:19.35 | *** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 19:20.05 | jbalcomb | [TK]D-Fender Executing NoOp("Zap/29-1", "") in new stack |
| 19:20.09 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:20.32 | [TK]D-Fender | jbalcomb : and its listed as DND in IPS? |
| 19:21.54 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:22.17 | jbalcomb | [TK]D-Fender when they are in Dnd status dnd/SIP/63:0800 exists, when they go off, either by hitting **63 on thier phone or click the checkbox in IPS, the family/key:value is DBDel'd from the database |
| 19:22.30 | cyber | how do i tell asterisk to passthru codecs without doing full proxy |
| 19:22.44 | r_evolution | type demons are killing me today [TK] |
| 19:22.44 | [TK]D-Fender | jbalcomb : Is it in DND status NOW? |
| 19:22.49 | r_evolution | TYPO... TY... PO |
| 19:22.49 | jbalcomb | [TK]D-Fender no |
| 19:23.02 | [TK]D-Fender | jbalcomb : then you are defeating the point of the test~! |
| 19:23.34 | jpablo | hey people, any article when i can read what dundi can do for me ? (practically) |
| 19:23.35 | [TK]D-Fender | jbalcomb : the noop is to verify that we can RETRIEVE the value properly when it IS set. |
| 19:23.40 | jbalcomb | [TK]D-Fender eh? the DnD feature works just fine. I'm trying to get rid of that WARNING |
| 19:24.14 | jbalcomb | [TK]D-Fender ive done a database show and seen the entry in there |
| 19:24.19 | [TK]D-Fender | jbalcomb : Pastbin the warning you currently get and the line causing it. |
| 19:25.14 | ManxPower | cyber, it will do that by default if it can. |
| 19:25.36 | ManxPower | cyber, many options to Dial (m, T, t, w, W, and maybe r) will prevent passthru |
| 19:25.51 | ManxPower | so will two legs of the call using different codecs |
| 19:27.14 | *** join/#asterisk svenna_ (n=svenna@p548D05B1.dip0.t-ipconnect.de) |
| 19:27.27 | jbalcomb | [TK]D-Fender http://pastebin.com/510292 |
| 19:27.58 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 19:28.00 | jbalcomb | [TK]D-Fender I apologize if my confusion or poor explanation has cause you distress. |
| 19:28.16 | jbalcomb | [TK]D-Fender you have been very helpful and i really do appreciate that |
| 19:28.58 | [TK]D-Fender | jbalcomb : use this instead - exten => s,4,GotoIf($[X${DB(dnd/SIP/63)}=X0800]?8) |
| 19:29.12 | [TK]D-Fender | jbalcomb : No distress, just trying to help. |
| 19:29.49 | [TK]D-Fender | I put the X back in like iCEBrkr's sample yesterday to avoid getting a blank on the left side of the "=". |
| 19:30.18 | [TK]D-Fender | jbalcomb : But we really need to check that value WHILE its set to DND. |
| 19:30.20 | Trazz | can you put something on pc to listen to all the gsm files? |
| 19:30.23 | iCEBrkr | jbalcomb: Are you causing problems again? |
| 19:30.42 | [TK]D-Fender | Trazz : Winamp seems to work fine, |
| 19:31.04 | jbalcomb | [TK]D-Fender is there anything to the fact that ixebrkr had the X before the first $? |
| 19:31.27 | iCEBrkr | jbalcomb: it's a PLACE HOLDER FOR THE 15th time! |
| 19:31.29 | [TK]D-Fender | jbalcomb : Typo, my bad... |
| 19:31.44 | jbalcomb | iCEBrkr oh heya there fella. yeah, im being confusing and difficult. haha.. screw you hippie. |
| 19:31.49 | iCEBrkr | lol |
| 19:31.53 | [TK]D-Fender | jbalcomb : actually, no its right |
| 19:31.55 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 19:31.58 | iCEBrkr | Some people never change. |
| 19:32.22 | jbalcomb | well, shit man, i dont know nuttin about nuttin, how can i be expect to know better? ;) |
| 19:32.34 | Trazz | TK , what do you use to make new gsm then ? |
| 19:32.43 | iCEBrkr | jbalcomb: haha |
| 19:32.49 | [TK]D-Fender | jbalcomb : the X is there in case the DB check returns NOTHING. you need to have SOMETHING on the left side of the "=" to compare against th right side (which always has that fixed value of 0800) |
| 19:33.03 | jbalcomb | [TK]D-Fender when they are in DnD they value of the entry is definitely 0800 |
| 19:33.05 | [TK]D-Fender | Trazz : You make a script in your dial-plan and record from your phone. |
| 19:33.23 | [TK]D-Fender | jbalcomb : Yes, but can the new DB function RETREIVE it properly? |
| 19:33.28 | iCEBrkr | jbalcomb: It's possible the DB() function will return NULL/Nothing/Empty |
| 19:33.44 | jbalcomb | iCEBrkr yes, thats what I finally GROCK' |
| 19:33.49 | jbalcomb | d this morning |
| 19:33.55 | [TK]D-Fender | jbalcomb : thats what we're trying to prove. It could be that the old DBGet wasn't implemented "right" and thats why it worked at all... |
| 19:33.57 | iCEBrkr | LOL |
| 19:34.06 | *** join/#asterisk ToTo (n=ToTo@host71-159.pool874.interbusiness.it) |
| 19:34.54 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 19:35.00 | frenzy | hey all |
| 19:35.07 | *** join/#asterisk reni (n=nubb@gateway.digium.com) |
| 19:35.07 | [TK]D-Fender | Keep in mind that MS *fakes* old bugs that they fixed because certain programs RELIED on their existance to function... |
| 19:35.19 | frenzy | can someone please help me with this notice.. |
| 19:35.19 | frenzy | Jan 17 19:33:56 NOTICE[7189]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
| 19:35.40 | frenzy | after dialing I get that for *some* destinations |
| 19:35.45 | frenzy | and call doesnt connect |
| 19:35.55 | jbalcomb | [TK]D-Fender That fixed it!!! |
| 19:36.04 | [TK]D-Fender | jbalcomb : If you say so... |
| 19:36.06 | [TK]D-Fender | :) |
| 19:36.16 | tzanger | frenzy: the destinations are using CNG. |
| 19:36.25 | [TK]D-Fender | frenzy : Make them stop! |
| 19:36.32 | jbalcomb | [TK]D-Fender well, /fixed/ is a funny way to look at it but that damn WARNING is gone. |
| 19:36.37 | iCEBrkr | haha |
| 19:36.52 | jbalcomb | iCEBrkr you told me wrong sucka. |
| 19:37.03 | frenzy | what are the pros and cons of that? |
| 19:37.13 | iCEBrkr | jbalcomb: I've been trying to tell ya to stuff that 'X' in there to make the expression evaluate and not spew the warning... Now if the DB() function was being called wrong that's a nother problem :) |
| 19:37.40 | iCEBrkr | jbalcomb: I don't think so.. Unless I was in a hurry to just spew an example. |
| 19:37.51 | jbalcomb | iCEBrkr yeah well i tried your X but it didnt work. [TK]D-Fender's X worked just fine though. |
| 19:37.58 | iCEBrkr | LOL |
| 19:38.04 | iCEBrkr | jbalcomb: I think you're stoned, but ok. |
| 19:38.04 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 19:38.21 | jbalcomb | iCEBrkr you had the X before the first $ but tk had his inside the [] before the 2nd $ |
| 19:38.52 | iCEBrkr | GotoIf("X$[${DIALSTATUS}]" != "X"]?h|1) |
| 19:38.57 | foucaulo | [TK]D-Fender: Thanks to help me with to solve my problem! you were right! that was the outbound channel who place calls to my voip provider..... |
| 19:39.02 | jbalcomb | iCEBrkr of course, if I had a clue I could have made that adjustment myseklf |
| 19:39.04 | iCEBrkr | jbalcomb: That's not where I had it? |
| 19:39.08 | [TK]D-Fender | My "X"'s make the best dialplans around and damn right they're better than yours... damn right they're better than yours.. I could teach you ... but I'd have to charge :D |
| 19:39.22 | iCEBrkr | [TK]D-Fender: LOL |
| 19:39.27 | jbalcomb | iCEBrkr yeah, thats like what you told me. it didn't work. :( |
| 19:39.27 | iCEBrkr | I need a better 'X' :( |
| 19:39.34 | iCEBrkr | jbalcomb: works in 1.2.0 |
| 19:39.44 | jbalcomb | ah |
| 19:39.46 | [TK]D-Fender | <iCEBrkr> GotoIf("X$[${DIALSTATUS}]" != "X"]?h|1) <- the X has to be INSIDE the test thus inside the [ |
| 19:40.00 | jbalcomb | iCEBrkr please keep your lousy deprecated code to yourself pal |
| 19:40.06 | iCEBrkr | [TK]D-Fender: It's prepended to the result of $[] tho |
| 19:40.11 | tzanger | iCEBrkr: hahaha |
| 19:40.18 | iCEBrkr | jbalcomb: You're deprecated |
| 19:40.25 | [TK]D-Fender | iCEBrkr : NO GOOD! Gotoif expects a 1 or a 0, thats it... |
| 19:40.29 | tzanger | I think you misspelled "ugly" |
| 19:40.34 | iCEBrkr | [TK]D-Fender: Neat. |
| 19:40.47 | iCEBrkr | GotoIf("$[X${DIALSTATUS}]" != "X"]?h|1) |
| 19:40.51 | iCEBrkr | So X marks the spot this time? |
| 19:41.06 | iCEBrkr | LOL |
| 19:41.09 | iCEBrkr | Guess what? |
| 19:41.11 | jbalcomb | iCEBrkr perhaps. I'm happy to have that WARNING gone. 3-6 times a SECOND!! |
| 19:41.12 | [TK]D-Fender | that one = much better |
| 19:41.16 | iCEBrkr | That's the only place it's wrong in my dial plan |
| 19:41.17 | iCEBrkr | hahaha |
| 19:41.20 | tzanger | bah just use my super duper hangup macro |
| 19:41.34 | tzanger | handles all dialstatus and hangupcauses and even does proper audio indications |
| 19:41.35 | iCEBrkr | I just did a search for my GotoIf's and it's correct in the rest of them. |
| 19:41.42 | tzanger | it's part of a balanced dialplan |
| 19:41.48 | jbalcomb | iCEBrkr "reciprocity mutha fucka!" |
| 19:42.15 | iCEBrkr | tzanger: What are you babbling about? |
| 19:42.23 | jbalcomb | hrmm.. how to make asterisk call iCEBrkr over and over with that wav.... |
| 19:42.46 | Trazz | tK, do you have that example script for recording ? |
| 19:42.58 | iCEBrkr | jbalcomb: I have 3 PRI available... AND I already have the basic war-dialer... >: | |
| 19:43.28 | [TK]D-Fender | Trazz : http://pastebin.ca/37104 |
| 19:43.37 | jbalcomb | iCEBrkr haha.. you still coding in Pascal? |
| 19:43.43 | iCEBrkr | jbalcomb: lol |
| 19:43.53 | iCEBrkr | jbalcomb: Shhhh Shut-up dude.. |
| 19:44.04 | jbalcomb | ok, i have to change that GotoIf statement in 53 more places.. bbiab. |
| 19:44.08 | [TK]D-Fender | jbalcomb : Whats wrong with that? I was m@d l33t in DOS with TP7! |
| 19:44.13 | jbalcomb | iCEBrkr oh right, its a secret. |
| 19:44.17 | iCEBrkr | [TK]D-Fender: haha me too |
| 19:44.25 | [TK]D-Fender | jbalcomb : I could reduce that for a minor charge :D |
| 19:44.29 | jbalcomb | You people are sick. |
| 19:44.29 | iCEBrkr | [TK]D-Fender: I wrote ShockWavE:PRO BBS in TP |
| 19:44.35 | Trazz | thanks |
| 19:44.52 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:45.15 | Beirdo | I was a Maximus BBS type personally |
| 19:45.29 | Beirdo | of course, knowing the author may have something to do with it |
| 19:45.35 | [TK]D-Fender | iCEBrkr : I ran WWIV and RemoteAccess |
| 19:45.40 | jbalcomb | [TK]D-Fender iCEBrkr Thank you both very much. I have 999,999,999 more issues though so drink up. |
| 19:45.53 | iCEBrkr | haha |
| 19:45.55 | Beirdo | wonder where Scott's working these days. |
| 19:46.05 | iCEBrkr | I ran WWIV for awhile, cuz no one else in the AC ran it |
| 19:46.29 | iCEBrkr | But then I got sick of all BBS software so I wrote my own. |
| 19:46.31 | Trazz | so Tk, just rename this to the name i want right (/tmp/asterisk-recording:gsm |
| 19:46.32 | Beirdo | BinkleyTerm + Maximus :) And did some Telegard in there too |
| 19:46.40 | justinu | wwiv! |
| 19:46.52 | file | Nugget: I saw Spinach Nuggets at the store yesterday! O.O |
| 19:47.10 | iCEBrkr | There were 100 Telegard boards.. 1000 Renegade boards.. Blah |
| 19:47.10 | Nugget | hah |
| 19:47.15 | harryvv | found a interesting feature that may not have been mentioned for AAh. play the vm back a second or more or then entire message with a single button. |
| 19:47.15 | [TK]D-Fender | file : I bought a pack from CostCo yesterday file :D |
| 19:47.24 | file | [TK]D-Fender: bleh |
| 19:47.29 | justinu | i liked telegard |
| 19:47.35 | file | pure insanity |
| 19:47.38 | harryvv | what is telegard? |
| 19:47.39 | [TK]D-Fender | file : Will let you know if they're any good... |
| 19:48.07 | iCEBrkr | justinu: Yea, I kinda stole Telegards menu config system |
| 19:48.29 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 19:48.53 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:49.08 | FuriousGeorge | my init.d/script for asterisk is no working so good and i never learned to manually start it ;) so now im having lunch with asterisk -cvvvv running in a remote console in front of me. hope my connection doesnt flake out |
| 19:50.10 | [TK]D-Fender | FuriousGeorge : Tried "safe_asterisk &" ? |
| 19:50.40 | *** part/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:50.43 | riddlebox | what is the easiest language to do AGI in? |
| 19:51.04 | justinu | probably perl or php |
| 19:51.08 | [TK]D-Fender | iCEBrkr : I wrote a WWIV user stats program that allowed me to drop to DOS on any system running it :) |
| 19:51.13 | Nugget | a language you already know. |
| 19:51.24 | iCEBrkr | [TK]D-Fender: lol |
| 19:51.32 | riddlebox | Nugget,I know python but it is not working so well |
| 19:51.42 | justinu | cnet was nice bbs software |
| 19:51.45 | justinu | only ran on Amigas tho |
| 19:51.54 | Trazz | pcboard |
| 19:52.01 | iCEBrkr | C=Net |
| 19:52.04 | justinu | pcboard was interesting too |
| 19:52.14 | iCEBrkr | I thought C=Net was a commie deal? |
| 19:52.17 | justinu | default setup sucked, but you could pimp it out pretty good |
| 19:52.23 | riddlebox | justinu,do you know of any perl tutorials for agi? |
| 19:52.30 | justinu | commodore made the amiga :) |
| 19:52.32 | justinu | sorta |
| 19:52.33 | Trazz | how about frontdoor :) |
| 19:52.37 | Trazz | fidomail |
| 19:52.46 | iCEBrkr | Fidomail was an abortion. |
| 19:53.00 | iCEBrkr | but I was able to make ShockWavE:PRO make Fidopackets |
| 19:53.18 | Trazz | we used to pack all this mail up and send it along via dialup.. those were the days |
| 19:53.37 | Trazz | and bbs doors |
| 19:53.48 | FuriousGeorge | [TK]D-Fender: like a charm :) im done with init.d/scripts forever |
| 19:57.44 | *** join/#asterisk Abbas (n=Abbas@203.81.193.166) |
| 19:59.03 | cyber | <PROTECTED> |
| 19:59.09 | cyber | why do i get this |
| 19:59.23 | [TK]D-Fender | justinu : I was a co-admin on a PCBoard once... kinda miss the old days.. |
| 19:59.34 | justinu | hehe |
| 19:59.36 | justinu | life was simpler :) |
| 20:00.16 | [TK]D-Fender | FuriousGeorge : I just call that from my rc.local |
| 20:01.35 | FuriousGeorge | [TK]D-Fender: my understanding of shell scripting is minimal at best. since i use gentoo dont i need to define a script for /etc/init.d/asterisk zap|status|start|stop, or at least all of those minus status maybe |
| 20:01.48 | FuriousGeorge | or *wouldnt i need to define scripts |
| 20:01.58 | r_evolution | ok so uhm... sqlmaestro + win32 = the suck |
| 20:03.55 | r_evolution | hey it's justinu my newest best friend with the cat who stalks me :-\ |
| 20:04.14 | [TK]D-Fender | FuriousGeorge : I know next to NOTHING and I run Slackware :) Just keep slapping in commands to rc.local and let the goot times roll! |
| 20:04.24 | r_evolution | haha @ ^ |
| 20:04.42 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 20:06.00 | eKo1 | Creating init.d scripts isn't that difficult. |
| 20:06.14 | eKo1 | Just take a working example and modify to your needs. |
| 20:06.24 | [TK]D-Fender | San someone confirm this for me : does Debian use the same run-level style scripts RH based distro's use? |
| 20:06.41 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 20:06.43 | justinu | i can't confirm, but I believe it's true |
| 20:07.01 | *** join/#asterisk ckruetze (n=ckruetze@i577A57DA.versanet.de) |
| 20:07.19 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 20:07.22 | r_evolution | for the most part D-Fen |
| 20:07.46 | *** join/#asterisk pwell (n=pimpwell@ool-18e41689.dyn.optonline.net) |
| 20:08.22 | Ayano | When I do a sip show peers, the status of an extension says 140ms, but if I ping the IP, it is less than a ms. What would cause that? |
| 20:08.56 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 20:08.57 | [TK]D-Fender | r_evolution : I just need to catchup to the more popular linux implementations. Slack has been good for me, but I need to be more "portable" |
| 20:09.50 | r_evolution | You should man... I dunno I got too used to RH for a while... and it got old so i started moving around |
| 20:10.27 | r_evolution | if you're looking for one to give to a current win user... SUSE seemed to have a pretty similar interface... is the one I got the sysadmin here to start diggin on *nix with |
| 20:10.42 | ckruetze | Could somebody give me a test call at christian@sip.balticfinance.com I'm somehow struggeling with my firewall |
| 20:11.06 | r_evolution | i kinda like debian... for now ;x but I hear you on the portability... as far as comfort levels go |
| 20:11.37 | pwell | I'm located in White Plains, NY. Looking to work with an asterisk engineer on a web based project in which we will be 50-50 partners. Contracts and all written up. I personally do PHP, HTML/CSS, Graphic Logo Design and Database work. I plan on making an automated system of some sort using precorded messages and a nice backend hooked in to the logs. If anyone is interested, let me know. My name is Larry |
| 20:12.13 | Nugget | http://www.livejournal.com/users/fantasygoat/69911.html <-- heh |
| 20:12.14 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 20:12.19 | [TK]D-Fender | ckruetze : Calling |
| 20:12.28 | frenzy | whats app_milliwatt.so used for? |
| 20:12.53 | ckruetze | [TK]D-Fender, thanks |
| 20:13.04 | r_evolution | ... |
| 20:13.08 | r_evolution | that's the best thing i've ever seen. |
| 20:13.22 | r_evolution | not really |
| 20:13.24 | [TK]D-Fender | ckruetze : no ringing, no nothing... its kinda dead |
| 20:13.25 | zoa | hey ho |
| 20:13.28 | r_evolution | but you know... it definitely get a smirk |
| 20:13.31 | zoa | anybody running an X desktop here |
| 20:13.32 | zoa | ? |
| 20:13.40 | Nugget | does OS X count? :) |
| 20:13.48 | zoa | hmm, not yet :) |
| 20:13.52 | zoa | maybe tomorrow |
| 20:14.21 | frenzy | ? |
| 20:14.21 | ckruetze | [TK]D-Fender: Our firewall guy says he can see UDP packages from you but he doesn't know why he is blocking them. |
| 20:14.31 | frenzy | whats app_milliwatt.so used for? |
| 20:14.35 | zoa | nugget, want to test an alpha of a mac softphone ? |
| 20:14.51 | mog_work | too annoy people frenzy |
| 20:15.02 | [TK]D-Fender | ckruetze : is your server behind NAT? |
| 20:15.04 | mog_work | its really usefull for testing things too |
| 20:15.10 | FuriousGeorge | [TK]D-Fender: i like how it keeps the colors when i connect to the safe_asterisk console with the -rvvv |
| 20:15.23 | Nugget | sure |
| 20:15.25 | frenzy | huh |
| 20:15.25 | frenzy | is there a way to go around CNG |
| 20:15.42 | ckruetze | [TK]D-Fender: Yes, it is, I wish it wasn't |
| 20:15.49 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 20:15.51 | [TK]D-Fender | ckruetze : Do you have either EXTERNHOST or EXTERNIP, and LOCALNET in SIP.conf? |
| 20:16.06 | ckruetze | [TK]D-Fender: all of them |
| 20:16.15 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 20:16.19 | [TK]D-Fender | ckruetze : And UDP is simply not making it to your * box? |
| 20:16.23 | VJ | Hello |
| 20:16.28 | ckruetze | [TK]D-Fender: The firewall guy said he allows the UDP stuff now, could you try again? |
| 20:16.32 | [TK]D-Fender | While believing things are forwarded properly? |
| 20:16.51 | VJ | in order to dial a number my telco provider needs the call to be sent on 5070 |
| 20:17.05 | VJ | i have changed the port number in sip.conf |
| 20:17.18 | VJ | is it required to be send somewhere else as well |
| 20:17.27 | [TK]D-Fender | ckruetze : Jan 17 15:27:32 WARNING[20411]: chan_sip.c:9527 handle_response_invite: Forbidden - wrong password on authentication for INVITE to |
| 20:17.53 | [TK]D-Fender | ckruetze : make sure you have "allowguest=yes" and a context for the call to land on as well... |
| 20:18.45 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 20:18.50 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 20:19.57 | *** join/#asterisk c3r3al-kill3r (n=Nbm@87-196-10-189.net.novis.pt) |
| 20:20.07 | ckruetze | [TK]D-Fender: Done, I added allowguest=yes in the context in extensions.conf, could you try once more, please? |
| 20:21.24 | *** join/#asterisk epablo (n=epablo@200.11.209.136) |
| 20:21.30 | epablo | hi people |
| 20:22.34 | epablo | I'm fighting with AMP and one of its agi's for a couple hours with no luck |
| 20:22.54 | fugitivo | ~amp |
| 20:22.55 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
| 20:23.19 | epablo | Sorry... thanks.. but I was going to ask a AGI specific question |
| 20:23.53 | fugitivo | ok :) |
| 20:23.59 | [TK]D-Fender | ckruetze : Same error |
| 20:25.30 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 20:25.31 | epablo | ANy good reason for $device = $AGI->database_get not to asign anything on $device, even when the result is AGI Rx << DATABASE GET AMPUSER 40/device, AGI Tx >> 200 result=1 (40) |
| 20:25.48 | ckruetze | [TK]D-Fender: Is allowguest=yes something newer? I'm using an older version of * at the moment? |
| 20:26.20 | [TK]D-Fender | ckruetze : Not AFAIK. pastebinyour [general] sectio of sip.conf |
| 20:26.24 | [TK]D-Fender | ~pb |
| 20:26.25 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:29.56 | ckruetze | [TK]D-Fender: http://pastebin.com/510405 |
| 20:30.02 | [TK]D-Fender | epablo : Have you upgraded your * setup from 1.0.x while keeping the same release of AMP? |
| 20:30.35 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 20:30.42 | epablo | I installed amp on 1.2.1 |
| 20:31.07 | [TK]D-Fender | ckruetze : And you've got a sipincoming context in extensions.conf with "exten => christian,1,something here" ? |
| 20:31.25 | [TK]D-Fender | epablo : Latest AMP I hope. |
| 20:31.57 | ckruetze | [TK]D-Fender: yes, but I just restarted * since it didn't like the SIP reload |
| 20:32.04 | fndude | I have a SIP server that I connect to for my VOIP, I am not able to recieve calls. I have SIP debug in CLI on, should I be able to "see" attempts from the server to connect to my asterisk box when I make a call to that number? |
| 20:32.16 | epablo | Fender: Version 1.10.010 |
| 20:32.18 | ckruetze | [TK]D-Fender: Could you try once more to call? |
| 20:32.42 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:32.48 | r_evolution | [TK] is handling it all |
| 20:33.00 | justinu | TK is a machine |
| 20:33.39 | r_evolution | i know! |
| 20:33.46 | r_evolution | http://www.syslog.com/~jwilson/pics-i-like/21_podborka_4.jpg = TK |
| 20:34.53 | justinu | lol |
| 20:35.14 | r_evolution | Nuggetz posted that earlier... |
| 20:35.26 | r_evolution | i plan to post it up at ANY available opportunity |
| 20:35.26 | *** join/#asterisk techie (i=gus@antibala.com) |
| 20:35.34 | *** join/#asterisk areski (n=areski@42.Red-83-34-9.dynamicIP.rima-tde.net) |
| 20:35.43 | r_evolution | "Oh... you're having problems with the T1 today?" you = http://www.syslog.com/~jwilson/pics-i-like/21_podborka_4.jpg |
| 20:35.54 | justinu | heh |
| 20:36.04 | lunk | haha |
| 20:36.09 | lunk | that pic owns, i think he's got it under control |
| 20:36.18 | fugitivo | i'm calling my telco, i'm onhold, and music sounds like if the cpu is at 100% |
| 20:36.18 | fugitivo | lol |
| 20:36.22 | *** join/#asterisk Octoploid (n=Octoploi@port-212-202-34-162.dynamic.qsc.de) |
| 20:36.38 | eKo1 | how does a cpu at 100% sound like? |
| 20:36.44 | zoa | choppy |
| 20:36.51 | troyb | cpu's make noise? |
| 20:37.02 | fugitivo | choppy and shitty |
| 20:37.19 | troyb | mine has never got any louder then a mute person |
| 20:37.22 | troyb | ohh wait :( |
| 20:37.35 | fugitivo | i wonder if they use asterisk |
| 20:37.41 | troyb | lol |
| 20:38.03 | Octoploid | Is there a way to switch to a white backgound (and black font) in console mode? |
| 20:38.15 | justinu | sun style |
| 20:38.17 | Dandan | it is a matter of your console |
| 20:38.19 | r_evolution | some days i call my brain and the moh sounds like the cpu is at 100% :-\ |
| 20:38.20 | Dandan | not asterisk |
| 20:38.25 | r_evolution | thats why i listen to dnb :-D |
| 20:38.30 | justinu | lol |
| 20:38.59 | [TK]D-Fender | ckruetze : same again |
| 20:38.59 | r_evolution | www.bassdrive.com justin :-D |
| 20:39.18 | Octoploid | Dandan: No it's not unfortunately... |
| 20:39.36 | r_evolution | you know what's bad? when the batteries on your wireless kb/mouse are dying... but you really just DONT feel like getting new ones... so you deal with keys constantly being missed :-\ |
| 20:39.42 | *** part/#asterisk epablo (n=epablo@200.11.209.136) |
| 20:39.59 | justinu | r_evolution: that's pretty lazy |
| 20:40.12 | fugitivo | this cpu 100% like moh is pissing me off... |
| 20:40.31 | r_evolution | eh. it is. |
| 20:40.44 | r_evolution | it's really that i just dont feel like doing it... and i keep hoping someone else will do it for me |
| 20:40.49 | *** part/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 20:41.04 | justinu | who's gonna replace that batteries in your own keyboard? |
| 20:41.15 | r_evolution | dont know : |
| 20:41.17 | r_evolution | :( |
| 20:41.19 | r_evolution | you? :-D |
| 20:41.21 | zoa | you know what is even worse ? |
| 20:41.27 | r_evolution | ? |
| 20:41.32 | zoa | if they died completely and you still dont want to get new batteries! |
| 20:41.35 | gaz00 | batteries in the keyboard? |
| 20:41.40 | r_evolution | :-D |
| 20:41.53 | r_evolution | copypaste using mouse ;x |
| 20:41.57 | zoa | yes |
| 20:42.00 | zoa | letter by letter |
| 20:42.04 | zoa | i do that sometimes |
| 20:42.04 | zoa | :) |
| 20:42.07 | gaz00 | just make a text file with every letter in it now |
| 20:42.09 | zoa | for special chars |
| 20:43.53 | ckruetze | [TK]D-Fender: This is the sip incoming part of my extensions.conf http://pastebin.com/510439 |
| 20:43.54 | fugitivo | i like playing dtmf music |
| 20:44.14 | _Sam-- | anyone have any opinions on IAX providers (besides teliax)? |
| 20:44.20 | _Sam-- | <hi zoa!> |
| 20:44.29 | zoa | hey ho sam! |
| 20:44.39 | _Sam-- | i cant get a good route for one of my clients to teliax, need another provider |
| 20:44.44 | [TK]D-Fender | ckruetze : Do an Answer before the dial... |
| 20:44.47 | _Sam-- | testing www.iax.cc right now, they seem ok |
| 20:45.43 | [TK]D-Fender | ckruetze : and remove the ",r" from your dial, its really not necessary |
| 20:45.44 | r_evolution | zoa is like a copy paste FIIIIIIIIIIIIIEND |
| 20:46.08 | r_evolution | zoa on keyboard = http://www.syslog.com/~jwilson/pics-i-like/21_podborka_4.jpg |
| 20:46.20 | zoa | i think the combination with t9 would be possible to type with |
| 20:46.21 | r_evolution | i told you... EVERY available opportunity |
| 20:46.27 | zoa | 9 keys with the mouse |
| 20:46.30 | zoa | t9 |
| 20:46.34 | r_evolution | O_o |
| 20:46.38 | zoa | grooooovy |
| 20:46.47 | ckruetze | [TK]D-Fender: Done |
| 20:47.02 | *** join/#asterisk scubes13 (n=klanders@cpe-065-184-093-063.sc.res.rr.com) |
| 20:47.08 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 20:47.26 | [TK]D-Fender | ckruetze : It answered, but died on your end this time |
| 20:47.32 | [TK]D-Fender | whats the CLI for the call say? |
| 20:47.47 | ckruetze | [TK]D-Fender: It says no such host :( |
| 20:47.48 | *** part/#asterisk scubes13 (n=klanders@cpe-065-184-093-063.sc.res.rr.com) |
| 20:47.58 | [TK]D-Fender | pastebin the call |
| 20:48.23 | r_evolution | >:O * + not connecting to mysql server = >:O |
| 20:48.56 | ckruetze | [TK]D-Fender: http://pastebin.com/510448 |
| 20:49.24 | [TK]D-Fender | ckruetze : Your SIP phone isn't set up right |
| 20:49.38 | [TK]D-Fender | # |
| 20:49.38 | [TK]D-Fender | <PROTECTED> |
| 20:49.38 | [TK]D-Fender | # |
| 20:49.38 | [TK]D-Fender | Jan 17 21:39:12 WARNING[13024]: chan_sip.c:1542 create_addr: No such host: 6398 |
| 20:49.45 | *** join/#asterisk beamer (n=nicl@c1-17-9.ndn.isadsl.co.za) |
| 20:49.46 | [TK]D-Fender | Stupd #'s.... |
| 20:50.03 | *** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 20:50.47 | r_evolution | UGH. |
| 20:50.48 | [TK]D-Fender | And hilarity, its using my work info routed through home :) |
| 20:50.49 | r_evolution | lunch. |
| 20:50.52 | r_evolution | away. |
| 20:50.53 | *** join/#asterisk elephantMan (n=elephant@149.202.99-84.rev.gaoland.net) |
| 20:50.55 | jbalcomb | [TK]D-Fender iCEBrkr all 55 entries have been updated. WARNING has been eliminated. Thank you both very much. |
| 20:51.18 | [TK]D-Fender | jbalcomb : remember only a small fee for me to completely renovate that mess for you ;) |
| 20:51.35 | FarrisG | A few of my users with Polycom 501s have reported callers complaining of sound popping in and out. I did a test call, and it sounds to me like there's some kind of noise gate - the signal seems to silence completely if the 501 user's voice goes below a certain volume. Is this most likely a 501 issue, or some sort of noise/echo cancellation settings on the asterisk server? |
| 20:51.35 | iCEBrkr | haha |
| 20:51.55 | iCEBrkr | [TK]D-Fender: rm -rf / |
| 20:51.59 | jbalcomb | [TK]D-Fender if only you weren't canadian. ;) |
| 20:52.13 | justinu | farrisg: i would suspect something other than the 501 |
| 20:52.15 | [TK]D-Fender | jbalcomb : Imperialist pig-dog! ;) |
| 20:52.27 | justinu | candians seem to own irc |
| 20:52.32 | justinu | nothing better to do |
| 20:52.41 | Charon212 | Does anyone know of a good free windows program to convert wav to gsm? |
| 20:52.45 | jbalcomb | FarrisG I am having a similar sounding issue on my Grandstream GXP-2000s |
| 20:52.59 | [TK]D-Fender | Charon212 : Use SOX in linux.... |
| 20:53.07 | iCEBrkr | Charon212: sox |
| 20:53.13 | zoa | http://www.asteriskguru.com/tools/audio_conversion.php -> i think wav conversion is fixed already |
| 20:53.22 | badboyz | is it possible to call up the telco and have your phone number released, so that your VOIP provider can provision it? |
| 20:53.30 | *** join/#asterisk jcims (n=jcims@12.28.112.46) |
| 20:53.31 | Charon212 | thanks |
| 20:53.41 | justinu | badboyz: you don't needc to call your telco |
| 20:53.47 | justinu | badboyz: the voip provider does it for you |
| 20:53.56 | badboyz | well i got a # w/ SBC right now |
| 20:53.59 | [TK]D-Fender | badboyz : Don't... if the provider supports LNP, let them start the takeover process or you could lose the # forever. |
| 20:54.05 | badboyz | i want to have it reassigned (not forwarded) to the voip provider |
| 20:54.17 | justinu | yeah, it should be possible |
| 20:54.27 | badboyz | so ask my voip provider to start a takover process? |
| 20:54.30 | Charon212 | Hmm, THat web utility isnt working. Whenever I try to use sox (Following the directions on the wiki) I recieve the error "sox: Premature EOF on .wav input file |
| 20:54.31 | Charon212 | " |
| 20:54.32 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 20:54.34 | justinu | yep |
| 20:54.35 | [TK]D-Fender | badboyz : exactly |
| 20:54.38 | badboyz | aok ty guys |
| 20:54.47 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:54.49 | cripito | hi |
| 20:54.56 | beamer | Howzit, |
| 20:55.19 | cripito | drumkilla 1 question.. the test with asterisk support silence suppression.. where can be downloaded? |
| 20:55.43 | cripito | did anyone known? |
| 20:55.50 | FarrisG | justinu: But this is only a problem on the 501s we have. My Grandstream GXP-2000 units all sound fine |
| 20:56.05 | justinu | farrisg: odd |
| 20:56.10 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 20:56.32 | iCEBrkr | Charon212: http://pastebin.ca/37114 |
| 20:57.31 | iCEBrkr | Charon212: That's the shell script I use to convert all my wav files |
| 20:57.39 | beamer | Anyone here done some integrations with legacy PABX's ? |
| 20:57.45 | Charon212 | thanks |
| 20:57.53 | Charon212 | lets see if that works |
| 20:57.57 | [TK]D-Fender | beamer : like? |
| 20:58.35 | cripito | yeap the question is like... |
| 20:58.52 | justinu | kinda in the middle of a pbx integration |
| 20:58.54 | beamer | Well, Im trying to get it connecting to an oldish PABX using a hfc card |
| 20:59.09 | beamer | Ill remember the name of it jusnow (long day) |
| 20:59.52 | pifiu | hello everyone |
| 21:00.08 | [TK]D-Fender | ckruetze : Got that SIP phone fixed? |
| 21:00.16 | beamer | Its my first bash, but I was wondering if there are any specific cable requirements |
| 21:00.42 | silentfury | has anyone configured an audiocodes gateway before? |
| 21:00.55 | beamer | I can make a call to the PABX but in reverse asterisk doesnt even see the incoming call |
| 21:01.00 | VJ | Hdllo |
| 21:01.02 | VJ | Hello |
| 21:01.09 | beamer | hoezit |
| 21:01.31 | VJ | Need to know how to send a call from sip.conf in registered mode |
| 21:01.33 | beamer | justinu: what hardware are you using ? |
| 21:01.37 | ManxPower | *whine* I don't want to go out in the rain |
| 21:01.42 | VJ | with a user name and password from behind the network |
| 21:01.43 | cripito | cheez... in less that 30 mins the trunk version went from 8139 to 8152 |
| 21:01.45 | cripito | :D |
| 21:01.55 | ckruetze | [TK]D-Fender: Yes, I think so, a Dial ... from the CLI now works. Could you try it once more? |
| 21:01.57 | *** join/#asterisk snova (n=oregonal@70.58.207.202) |
| 21:02.08 | znoG | can one do a logical AND in the dialplan with a GotoIf? like: GotoIf($["${ARG1}" = "1" & "${ARG2}" = "2"]label1:label2) ? |
| 21:02.15 | snova | anyone have any experience with VoIP to GSM gateways? |
| 21:02.17 | *** join/#asterisk guigouz (n=guigouz@201.6.252.170) |
| 21:02.31 | iCEBrkr | yay! My first tech support email for iBCallerID |
| 21:02.33 | justinu | beamer: toshiba strata |
| 21:02.33 | iCEBrkr | :-| |
| 21:02.43 | [TK]D-Fender | ckruetze : There we go... |
| 21:02.55 | justinu | we're having trouble getting the tie trunk up |
| 21:03.04 | guigouz | If I sign up to some SIP provider and configure asterisk properly, will everybody on my office be able to call (simultaneously) other numbers ? |
| 21:03.10 | justinu | well, actually the tie trunk is up, but we can't seem to make it ring extensions |
| 21:03.16 | ManxPower | znoG, README.variables is your friend. |
| 21:03.29 | beamer | What cards do you have in the Asterisk box ? |
| 21:03.32 | ckruetze | [TK]D-Fender: Thanks a lot, now that the problem is solved, I can finally attend my fiancée. See you around |
| 21:03.38 | justinu | 2 span T1 card |
| 21:03.41 | justinu | TE210? |
| 21:03.44 | VJ | hi |
| 21:03.45 | justinu | something like that |
| 21:03.49 | znoG | ManxPower: is that a yes its possible but you did it wrong? |
| 21:03.56 | cripito | manxpower did u hear anything about an attend to run asterisk with silence support? |
| 21:04.19 | znoG | ManxPower: in the wiki i read up on the GotoIf cmd and from what I understood its right. |
| 21:04.26 | *** part/#asterisk ckruetze (n=ckruetze@i577A57DA.versanet.de) |
| 21:04.37 | snova | any recommendations on VoIP to GSM gateways I should take a look at? Looking for an easy way to not eat up minutes by using Mobile to Mobile calling |
| 21:04.38 | VJ | need some basic support on sip.conf |
| 21:04.48 | beamer | justinu: ah, Im using a BRI card, do you know if you need to use a xover or something between the PABX and ISDN card ? |
| 21:05.29 | justinu | BRI isn't something I'm very familiar with, but there's T and U interfaces, I believe |
| 21:05.38 | justinu | maybe you need an NT1 to convert? |
| 21:06.28 | beamer | Justinu: Possibly, I got it ringing on the PABX side but the call dies after a few rings |
| 21:06.36 | harryvv | snova is this international calls? |
| 21:06.52 | ManxPower | znoG, That's a "read the damn file and find out" |
| 21:07.17 | snova | no... just normal cell phone calls |
| 21:07.20 | ManxPower | README.variables is the right place to go for this information, not the Wiki |
| 21:07.32 | *** join/#asterisk Druken (n=blowme@static.abss.ca) |
| 21:07.45 | justinu | beamer: not really sure... BRI is kinda complicated |
| 21:07.53 | cripito | :((((( i want my asterisk runing with silence supression snif snif... |
| 21:08.14 | harryvv | snova, I really dont think there is any other way. one idea if you are stationary in a area for a long period of time is go wifi voip phone. |
| 21:08.31 | cripito | and the patches don't run .. and the svn was moved :~-((( |
| 21:08.41 | beamer | justinu: cool, just what I wanted to hear :-) |
| 21:09.05 | harryvv | snova, make sence? |
| 21:09.17 | justinu | beamer: i don't think it's an NT1 issue if it works at all |
| 21:09.51 | guigouz | anyone use broadvoice or any other voip provider ? |
| 21:10.12 | snova | harryvv: actually I am trying to make call to a cell phone so they don't get charged minutes for receiving, and allowing them to call my PBX for free also and have the PBX dial out for them. |
| 21:10.48 | snova | most cell phone companies in the US make mobile to mobile minutes free. |
| 21:10.50 | beamer | Im worried bout the cable though, straight through doesnt work at all. Anyone know of a referance I can look at ? |
| 21:11.08 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 21:11.10 | justinu | beamer: BRI is 2 wire, right? |
| 21:11.42 | snova | harryvv: so using a VoIP to GSM bridge will allow free calls in and out the providers GSM network |
| 21:11.44 | beamer | nah 4 wire |
| 21:12.00 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 21:12.01 | justinu | oh, one of the interfaces is 2 wire |
| 21:12.34 | rculp | silly question, but does the voicemail config support html e-mail for the e-mail messages? |
| 21:12.39 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.73) |
| 21:12.41 | jbalcomb | iCEBrkr why converting wav to gsm? |
| 21:12.44 | snova | harrvv: so people can get into VoIP and the VoIP to GSM will call someone on the providers GSM network without charging the reciever minutes |
| 21:12.48 | beamer | Mmm, its one port, 4 wire's, 3 channels |
| 21:12.57 | snova | harryvv: any of this many sense? |
| 21:13.06 | justinu | yeah,here in the states, the telco was delivering 2 wire BRi |
| 21:13.18 | beamer | 2B channels (for voice) 1 D channel for control or something |
| 21:13.41 | dasenjo | Hi! Im getting 060117-161210 ERROR[22726]: chan_zap.c:6260 mkintf: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling |
| 21:13.42 | dasenjo | 060117-161210 ERROR[22726]: chan_zap.c:9200 setup_zap: Unable to register channel '1-31' |
| 21:13.44 | beamer | ah, I seem to have a lot of fun looking for support on BRI |
| 21:13.54 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:13.58 | dasenjo | but the zaptel anda zapata conf did not change |
| 21:14.07 | dasenjo | I need help! please! |
| 21:14.11 | rculp | and another question: we're receiving a lot of echo and chipmunk impressions using sip phones and the one hard phone we have for testing is a gxp2000 which is a peice of junk. Do you think the echo is softphone related ordue to the tdm wildcard 110 |
| 21:15.08 | snova | Wow! I think I actually had a question someone was not able to answer around here.. must be a first for me |
| 21:15.12 | cripito | ij i got a question about svn anyone that could help? |
| 21:15.33 | justinu | softphones are nortorious for echo |
| 21:16.09 | rculp | justinu: ok, figured |
| 21:16.57 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:16.59 | justinu | make sure your phones are configured to use 20ms RTP packets |
| 21:17.12 | rculp | and that's on the softphone side |
| 21:17.17 | justinu | yeah |
| 21:17.20 | VJ | Need some support configuring sip.conf |
| 21:17.22 | snova | ok.. anyone have any tricks for make and receive cell phone calls (in the US) without eating up minutes? besides my Mobile to Mobile using a VoIP to GSM gateway approach? |
| 21:17.42 | rculp | ok, thanks justinu |
| 21:17.47 | zoa | Snova, use a softphone with bluetooth |
| 21:17.48 | *** join/#asterisk zamsler_ (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 21:17.59 | zoa | and you wont need to buy a gsm gateway |
| 21:18.08 | VJ | my PSTN provider needs to get the calls with a user name and password |
| 21:18.18 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:18.19 | tainted_ | zoa do u have experience with gsm gateways? |
| 21:18.54 | cripito | hi zoa |
| 21:19.33 | rculp | anyone have an answer to getting voicemail.conf to sending html e-mails? |
| 21:19.35 | syle | whats a gsm gateway or some bluetooth device gonna do when i think hes talking about say heading to caesars palace for the weekend with his cell phone and wanting to make calls from the casino |
| 21:19.46 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 21:19.50 | snova | zoa: I think I might not have been clear... I want to be able to use a cell phone where IP is not available. |
| 21:19.58 | Druken | anyone know if there's a SIP FXO gateway? nice and cheap one... say i want to grab my home line and send it to me office pbx? :) |
| 21:20.01 | zoa | aaaah |
| 21:20.05 | guigouz | if I have one voip provider, but just one account, can I make multiple calls (SIP) simultaneously ? |
| 21:20.14 | zoa | guigouz: depends on the voip provider |
| 21:20.31 | justinu | if you can't, it's an artificial limit :P |
| 21:21.12 | fndude | I can't seem to get incomming calls to work correctly. I have the sip client for my VOIP provider added as a pier and a user, and registered. I also have an entry in extensions.conf with the number they gave me, but no love. arg. |
| 21:21.25 | [TK]D-Fender | Druken : Plenty of them |
| 21:21.33 | snova | syle: you are on the right track.. using the cell phone to call my GSM gateway so it can get me onto VoIP |
| 21:21.42 | justinu | Druken: SPA-3000... $100 |
| 21:21.43 | *** join/#asterisk lorinc (n=ang@caracas-2902.adsl.interware.hu) |
| 21:21.55 | *** join/#asterisk IRS (n=Nbm@87-196-46-124.net.novis.pt) |
| 21:22.14 | Druken | ahh, but i've heard alot of people having issues.... |
| 21:22.17 | Druken | like getting CID |
| 21:22.27 | snova | cell phone call would be make out in the middle of nowhere. |
| 21:22.33 | justinu | hmm, i haven't really had a chance to try mine out |
| 21:22.40 | justinu | hard to find a POTS line these days |
| 21:22.43 | FuriousGeorge | i got a couple of asterisk boxes running, and for some reason when there is no cid, rather than displaying unavailable one of them picks the caller id from one of the extensions ("Maria" 103@externip, in this case) the other one says "Asterisk 0@10.0.0.10" or something |
| 21:22.45 | *** join/#asterisk epablo (n=epablo@200.11.209.136) |
| 21:22.51 | [TK]D-Fender | Druken : CID problems are more often reported for channel-banks, not SIP-FXO gateways |
| 21:23.14 | Druken | [TK]D-Fender: do you have a spa-3000 ?? :) |
| 21:23.24 | [TK]D-Fender | Druken : thats what I use at home |
| 21:23.29 | FuriousGeorge | and mine are pots lines |
| 21:24.00 | epablo | Hi I've got a TDM400P with FXO modules. I'm looking for a way to make my IVR answer on the first ring. Right now it does it on the second ring |
| 21:24.01 | guigouz | zoa: I'm getting started with voip, was about to sign up with broadvoice, but just saw they only allow 1 device per account .. |
| 21:24.22 | *** join/#asterisk silentfury_ (n=a@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 21:24.30 | Druken | epablo: usecallerid=no |
| 21:24.32 | guigouz | zoa: even if I setup asterisk, I'll only be able to make one call at a time, right ? |
| 21:24.43 | epablo | OK.. let me test that.. |
| 21:24.53 | Druken | [TK]D-Fender: i will have to look at one of those then.... :) |
| 21:24.54 | snova | guigouz: I was looking into broadvoice.. but it started to look a little unreliable. |
| 21:25.13 | epablo | Druken: Thats on the zapata.conf right? |
| 21:25.13 | [TK]D-Fender | Druken : How many lines do you require? |
| 21:25.22 | guigouz | snova: I'm still testing all the options. do you recommend any other ? |
| 21:25.47 | Druken | [TK]D-Fender: well, i dunno... i'm just getting tired of my peice of shit TDM card crapping out on me |
| 21:25.51 | *** part/#asterisk beamer (n=nicl@c1-17-9.ndn.isadsl.co.za) |
| 21:25.51 | ManxPower | FuriousGeorge, for FXO ports use "as received" |
| 21:25.53 | ManxPower | FuriousGeorge, for FXO ports use "asreceived" |
| 21:25.54 | Druken | so i'm looking for other options |
| 21:26.21 | snova | guigouz: no first hand experience yet.. just reading forums on dslreports.com before I make my decision. |
| 21:26.24 | guigouz | snova: If I have any provider which allows one device per account, even if I setup asterisk, I'll only be able to make one call at a time ? |
| 21:26.36 | [TK]D-Fender | Druken : You don't know how may ports you need? |
| 21:27.07 | Druken | [TK]D-Fender: well, it would depend on how many lines i wanted to bring in... currently i have 3 analog lines i bring in |
| 21:28.03 | jbalcomb | Who wants to look at my 'Unable to handle indication 3' problem? http://pastebin.com/510521 |
| 21:28.11 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 21:28.44 | FuriousGeorge | ManxPower: "as received" where, in zapata.conf? callerid=asreceived <--likethat |
| 21:29.01 | ManxPower | FuriousGeorge, correct |
| 21:29.04 | Druken | yep |
| 21:29.31 | Druken | when no CID is given on my TDM card, i get asterisk as CID |
| 21:29.36 | FuriousGeorge | ManxPower: can i do that for the fxs so it displays "Name <101>" vs "Name <101@10.0.0.10> |
| 21:29.43 | *** part/#asterisk epablo (n=epablo@200.11.209.136) |
| 21:30.30 | snova | guigouz: I have heard good things about MyPhoneCompany and QuantumVoice.. also looking at Tellax |
| 21:30.57 | [TK]D-Fender | Druken : Get a Mediatrix or AudioCodes gateway then. |
| 21:30.58 | VJ | hi |
| 21:31.16 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:31.16 | VJ | need to dial out a number in registered mode |
| 21:31.20 | generalhan | whats going on everyone ?> |
| 21:31.56 | generalhan | anyone know where i can get the TE210 drivers, besides at ftp://ftp.digium.com/pub/zaptel/ ??? the FTP is down right now i think |
| 21:32.09 | [TK]D-Fender | generalhan : SVN |
| 21:32.14 | snova | guigoiz: correction teliax |
| 21:32.28 | *** join/#asterisk _tdonahue (n=tdonahue@208.51.101.201) |
| 21:32.31 | generalhan | [TK]: SVN ??? |
| 21:32.42 | [TK]D-Fender | generalhan : what replaced the old CVS |
| 21:32.44 | *** join/#asterisk backblue (n=moo@87-196-34-198.net.novis.pt) |
| 21:33.02 | generalhan | hmm |
| 21:33.04 | VJ | hi |
| 21:33.14 | generalhan | all i have loaded is the release 1.2.1 |
| 21:33.36 | VJ | anyone to guide me through the configuration of dialplan in sip.conf |
| 21:34.34 | VJ | anybody??????????????????????? |
| 21:34.36 | FuriousGeorge | just noticed that when i reload chan_zap.so i get Jan 17 16:34:06 WARNING[6572]: chan_zap.c:10816 setup_zap: Ignoring signalling |
| 21:34.36 | FuriousGeorge | <PROTECTED> |
| 21:34.40 | [TK]D-Fender | VJ : Diaplan = extensions.conf, not sip.conf |
| 21:34.47 | FuriousGeorge | but everything seems to work |
| 21:35.13 | VJ | i need to dial the numbers via VOIP |
| 21:35.27 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 21:35.44 | generalhan | VJ: he's just saying that you worded it incorrectly, if you are referring to your dialplan, that is extensions.conf, not sip.conf |
| 21:35.55 | jpablo | FuriousGeorge, as i remember reloading changed in zapata signaling don't work |
| 21:35.59 | VJ | i was able to dial the numbers successfully, when my VOIP provider was authenticating me using the prefix which i configured in extensions.conf |
| 21:36.42 | VJ | but now i have changed my vendor, he is saying that he can accept the call only when i will authenticate it using a user name and password he has given me |
| 21:36.52 | snova | for anyone who is interested what the heck I am babling about VoIP to GSM Gateway.. like this thing: http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html |
| 21:36.54 | VJ | i configured that username=user |
| 21:37.07 | VJ | and secret=password |
| 21:37.13 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 21:37.17 | snova | see the network diagram on the web page. |
| 21:37.35 | generalhan | VJ: when i was in your shoes trying to figure it all out in the begining i would do a pastebin on my entire extensions.conf and iax.conf, it will be much easier for you to get help if you can direct people to a pastebin that is waiting for them |
| 21:37.38 | FuriousGeorge | jpablo: so its probably just telling me that its ignoring the signalling cuz i aint doin a full restart |
| 21:38.00 | jpablo | FuriousGeorge, yes |
| 21:38.09 | VJ | i am doing that |
| 21:38.12 | FuriousGeorge | cool |
| 21:39.06 | r_evolution | something about the name 'FuriousGeorge' makes me go O_O |
| 21:39.22 | generalhan | lol |
| 21:39.56 | snova | CuriousBob? |
| 21:39.56 | FuriousBob | heh ;] |
| 21:40.02 | mog_work | FuriousBob,!?!! |
| 21:40.04 | FuriousBob | haha |
| 21:40.16 | FuriousBob | it's something thats just amusing as hell |
| 21:40.37 | Furious | damn |
| 21:40.40 | Furious | so much for variables :( |
| 21:40.46 | Furious | Furious(${name}) |
| 21:41.18 | generalhan | Can someone help me understand SVN ?? i need to get the digium drivers for the TE210 and the FTP site is not working for me right now |
| 21:41.36 | snova | generalhan: do you understand CVS? |
| 21:41.50 | generalhan | nope ive never used CVS before |
| 21:41.55 | ManWithYellowBat | has anyone seen my chimp> :) |
| 21:42.10 | brookshire | does http://ftp.digium.com work? |
| 21:42.26 | [TK]D-Fender | generalhan : SVN is a source distribution and versioning tool like CVS and uses its own repositories to hold multiple versions of the code. Odds are your distro has teh tools required to download the source direct from these resources. |
| 21:42.41 | [TK]D-Fender | Check out the WIKI on how to use them. |
| 21:42.49 | *** join/#asterisk wspraul (n=wspraul@dbl.q-ag.de) |
| 21:43.34 | generalhan | [TK] & everyone: sorry, i didnt even realize that the wct2xxp was a part of the zaptel-1.2.1.tar file. i have had it this whole time -- sorry ! |
| 21:43.39 | [TK]D-Fender | ok, I'm off, later all |
| 21:43.43 | harryvv | ManWithYellowBat ? |
| 21:43.44 | generalhan | Hasta |
| 21:44.11 | generalhan | so once i have the file i just need to do a `modprobe wct2xxp` and thats it ? |
| 21:44.28 | wspraul | wholy crap - so many people here. I'm browsing the www for hours but cannot find help: just setup asterisk, starts fine (I'm in CLI right now). Connected to my ISDN BRI (chan_capi works fine). Now I want to connect to the Asterisk box using H323 then from there out via ISDN. Any good URL with config examples? |
| 21:44.30 | FuriousGeorge | harryvv: b/c the man with the yellow hat was the companion of curious george the monkey. so its a play on words of sorts |
| 21:44.40 | *** join/#asterisk secure75 (n=mic@dslb-084-057-039-156.pools.arcor-ip.net) |
| 21:44.47 | *** join/#asterisk johnnyb (n=johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 21:44.55 | pifiu | wasup everyone |
| 21:44.57 | harryvv | okay |
| 21:45.04 | FuriousGeorge | CuriousGeorge:ManWithYellowHat::FuriousGeorge:ManWithYellowBat |
| 21:45.05 | harryvv | what ever makes sence to him |
| 21:45.25 | johnnyb | Is there a good tutorial on how to wire phone lines to a dmarc? I've only done asterisk installs where the phone cables were already connected. |
| 21:45.33 | FuriousGeorge | guess youve never heard of him |
| 21:46.11 | justinu | Johnnyb: it's 2 wires, red and gree |
| 21:46.17 | justinu | what's so hard about it? |
| 21:46.24 | FuriousGeorge | johnnyb: its easy, get an old phone and some aligator clips and you got a "butt sett" clip ono the pins till you get a dialtone, then call your cell and check your cid |
| 21:46.27 | *** join/#asterisk skamp|afk (i=skambar@p548439E2.dip0.t-ipconnect.de) |
| 21:46.37 | *** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 21:46.51 | FuriousGeorge | if you use verizon you may be able to dial 958 or something to have your number read back to you |
| 21:47.02 | johnnyb | justinu: I've just never done it. I also have a DSL line I need to install, and wasn't sure if there was anything special. We're moving from a rented building to a new one. |
| 21:47.15 | justinu | nah, dsl is just a normal phone line... except for the filters |
| 21:47.24 | wspraul | anybody using Asterisk to bridge between H323 and ISDN (capi)? |
| 21:47.51 | johnnyb | I've watched telco people do it, and they have a wierd tool for punching down lines, and I wasn't sure (a) where to locate such a tool, (b) what it actually does, and (c) which wire goes where. |
| 21:47.53 | FuriousGeorge | nothing special at all about it |
| 21:48.25 | wizhippo | any idea why my asterisk console isn't highlighting in color? Using putty |
| 21:48.40 | *** join/#asterisk TK9 (n=Administ@p54B2ABE0.dip0.t-ipconnect.de) |
| 21:48.42 | FuriousGeorge | you can use needle nose pliers and a screw driver, but you can get a free plastic tool with a modular jack purchace at home dePot |
| 21:48.44 | *** part/#asterisk TK9 (n=Administ@p54B2ABE0.dip0.t-ipconnect.de) |
| 21:49.21 | johnnyb | FuriousGeorge: interesting. Is there a tutorial with pictures on the web? Bits I can deal with, physical equipment I'm not so good with :) |
| 21:49.26 | rob0 | I was a big fan of Curious George back before the Epoch :) |
| 21:49.55 | wunderkin | rob0, before dinosaurs roamed the earth? |
| 21:50.18 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 21:50.19 | FuriousGeorge | johnnyb: there may be, but if you lack the propper tools its a good thing you only gotta do one line. its as easy as it sounds, get the wire inbetween the metal teeth, |
| 21:50.26 | FuriousGeorge | two teeth = one line |
| 21:50.32 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:50.42 | FuriousGeorge | you just gotta find which two = your dsl line |
| 21:50.43 | rob0 | before 1970/01/01 00:00 UTC of course :) |
| 21:50.50 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 21:51.03 | johnnyb | Aren't there two wires? Does it matter which is which? |
| 21:52.19 | justinu | try and be consistent |
| 21:52.51 | generalhan | Does anyone have a link for the wire locations for a T-1 cable? i just need to make a cable from my T-1 [out] to my Digium card [in], and im not sure it they use a straight through cable |
| 21:53.17 | FuriousGeorge | johnnyb: usually two teeth verticle of eachother are one line, going horizontal you can daisy chain across the remaining pins |
| 21:53.42 | wunderkin | generalhan, standard ethernet cable |
| 21:53.58 | generalhan | really?? just a straight though cable huh ? that suprises me |
| 21:54.00 | generalhan | thanks ! |
| 21:54.14 | justinu | if it's from network to CPE, yes |
| 21:54.18 | FuriousGeorge | if columns are letters and rows are numbers, try to put an incomming line on A1 and A2, then you can pick it up from B1 and B2 or C1 and C2 or even B2 and D1 if you want |
| 21:54.19 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 21:55.08 | FuriousGeorge | johnnyb: thats to you |
| 21:56.13 | justinu | any of you guys know how a e&m wink TIE trunk works on a traditional pbx? |
| 21:56.52 | shmaltz | justinu, what do you mean by how? |
| 21:57.12 | shmaltz | justinu, it should be RBS which means you get 24 channels vs 23 for pri |
| 21:57.31 | justinu | well, i place calls on this trunk, and outpulse the digits of the station i want to dial, right? |
| 21:57.40 | shmaltz | justinu, and it uses inband singalling for the DID at least, not sure about other things if its inband or not |
| 21:57.54 | justinu | i should get a wink after I outpulse, or before? I forgot! |
| 21:57.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:57.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:58.04 | shmaltz | justinu, don't know |
| 21:58.23 | justinu | i need to talk to a traditional pbx guy... drat |
| 21:58.24 | gaz00 | anyone gotten paging to work on an Asterisk PBX? |
| 21:59.07 | mog_work | what doesnt work gaz00 |
| 21:59.12 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 21:59.31 | generalhan | Can some one take a look at this for me ?? http://generalhan.pastebin.ca/37127 :: i JUST ran ztcfg and this is what i got ... am i ready to rock and roll ? or just roll on the ground crying ? |
| 21:59.53 | justinu | generalhan: cat /proc/zaptel/1 |
| 22:00.38 | r_evolution | sometimes... i wonder if FuriousGeorge has too much FuriousTime |
| 22:00.39 | r_evolution | :) |
| 22:00.41 | justinu | so far so good tho |
| 22:00.53 | generalhan | justinu: they all say TE2/0/1/[1-23] Clear, |
| 22:01.02 | justinu | first line |
| 22:01.12 | generalhan | Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" B8ZS/ESF ClockSource |
| 22:01.18 | justinu | looks ok then |
| 22:01.29 | generalhan | justinu: thanks a lot ! im gonna give it a try ! |
| 22:02.46 | gaz00 | i'm trying to test a setup and my softphones are not automatically picking up |
| 22:02.46 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:03.16 | gaz00 | calling one extension is routing properly, but the auto-answer isn't. |
| 22:03.27 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 22:03.36 | gaz00 | i wanted to know if softphones were handled differently than physical VoIP phones |
| 22:03.44 | justinu | nope |
| 22:04.02 | justinu | well, each type of SIP phone has a different way of doing auto-answer usually |
| 22:04.17 | gaz00 | hrm. |
| 22:04.22 | *** join/#asterisk Till (n=Miranda@p54B2ABE0.dip0.t-ipconnect.de) |
| 22:04.25 | gaz00 | that would be an issue then. I'll look into that. |
| 22:05.04 | gaz00 | second question: what about an automatic door lock? I.e. pushing a button to unlock the front door? |
| 22:06.18 | *** join/#asterisk sandynet (n=darktige@12.111.165.10) |
| 22:06.24 | fugitivo | gaz00: you need an adapter for that |
| 22:06.37 | *** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
| 22:06.43 | gaz00 | I have an electronic lock already. |
| 22:06.51 | pifiu | I have a question regarding a suggestion maybe you guys can help me out on |
| 22:07.14 | fugitivo | gaz00: you need a fxs or fxo adapter for the door |
| 22:07.27 | shmaltz | pifiu, this is IRC, not your 2nd grade class room |
| 22:07.36 | fugitivo | gaz00: check www.2n.cz |
| 22:07.40 | pifiu | lol |
| 22:07.43 | pifiu | thanks shmaltz |
| 22:08.07 | pifiu | I have 2 locations, each with their on asterisk machine on each end. One has an IAX provider, and the other has nothing yet |
| 22:08.19 | pifiu | i want to make calls from the second location that currently is not setup for anything |
| 22:08.33 | pifiu | should I make it route calls to the first location and then to the provider? |
| 22:08.38 | Ariel_ | pifiu, yes you can have both boxes talk to each other |
| 22:08.40 | gaz00 | hmm...and thank you fugitivo! |
| 22:08.43 | badboyz | so i just setup the latest version of * .. i can dialout and i can hear people talking, but they cant hear me -- whats the general problem with this? |
| 22:09.02 | justinu | why can't both boxes connect to the same IAX provider? |
| 22:09.09 | badboyz | and when i say dialout -- i mean, dialing to a voip provider -> pots/cell/etc |
| 22:09.30 | badboyz | its like my phone is muted |
| 22:09.45 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 22:10.09 | ManxPower | as long as you don't have to have incoming calls (and so don't have to register) you should just be able to send calls to the provider using the same auth info. |
| 22:10.20 | pifiu | ariel sent you a reply back to that PM you sent me |
| 22:10.28 | ManxPower | As long as your provider allows multiple calls, and I don't know of any per min plans that prohibit that. |
| 22:10.56 | pifiu | allows 5 outgoing port max at a time |
| 22:11.08 | pifiu | i have the IAX registration |
| 22:11.17 | pifiu | but does IAX limit you to one machine in use at a time? |
| 22:11.30 | Ariel_ | pifiu, no |
| 22:11.31 | justinu | no |
| 22:11.34 | pifiu | or is that dependaant ont he provider? |
| 22:11.40 | Ariel_ | provider only |
| 22:11.48 | pifiu | oh so i can have multiple machines connecting to the same IAX |
| 22:12.09 | Ariel_ | Not really |
| 22:12.25 | Trazz | does anyone have cisco 7940 working with * ? |
| 22:12.28 | ManxPower | pifiu, as long as it's outgoing only and the provider allows it. |
| 22:12.47 | Ariel_ | bbl I have to head home... |
| 22:12.50 | ManxPower | If you have to register => with the provider then this prolly won't work, of course. |
| 22:13.26 | *** part/#asterisk javar (n=javar@69.79.217.237) |
| 22:14.52 | pifiu | yeah i have to register |
| 22:16.05 | *** join/#asterisk c3r3al-kill3r (n=Nbm@87-196-36-106.net.novis.pt) |
| 22:17.01 | *** join/#asterisk brooks (n=nubb@gateway.digium.com) |
| 22:19.08 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:19.49 | r_evolution | Manx... here's a good one... the switch we're using as our backbone... must not have something set up correctly |
| 22:20.07 | *** join/#asterisk gaspiz (i=gaspi@86.34.6.164) |
| 22:20.18 | r_evolution | b/c i've had like 6 different incoming on the same SIP... |
| 22:20.30 | r_evolution | but then again... seeing as how it is OUR switch... sorta... i guess that helps :p |
| 22:21.14 | *** join/#asterisk Hmmhesays (n=Neg@72.24.227.83) |
| 22:21.53 | gaspiz | when a call is hang up during an agi(php) script, what signal does asterisk send to it? |
| 22:22.04 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:22.33 | gaspiz | i'm caching SIGTERM |
| 22:22.36 | mog_work | ahhhhhh |
| 22:23.25 | r_evolution | O_o http://www.syslog.com/~jwilson/pics-i-like/21_podborka_4.jpg |
| 22:23.27 | Hmmhesays | anyone ever heard of lignup? |
| 22:24.10 | SkramX | lignup? |
| 22:24.23 | *** join/#asterisk fa_back (i=faceoff@devel.acdbddh.eu.org) |
| 22:24.37 | r_evolution | you know... i looked too quickly... and i swear it said |
| 22:24.39 | *** part/#asterisk telmich (i=telmich@gpm/telmich) |
| 22:24.41 | r_evolution | "SkramX goggles" |
| 22:24.48 | Trazz | does anyone have cisco 7940 working with * ? |
| 22:25.03 | SkramX | Trazz: my business partner is sending me some in a week, I cant wait. |
| 22:25.09 | SkramX | Are you running SIP or SCCP on it./ |
| 22:25.10 | SkramX | *? |
| 22:25.15 | Trazz | i am running SIP on it |
| 22:25.20 | SkramX | Hmm |
| 22:25.28 | SkramX | Hmmhesays: what about lignup? |
| 22:25.29 | Trazz | but need to know how to get it running |
| 22:26.02 | SkramX | Welp, do you have it setup to register with asterisk, and will asterisk accept the registration request? |
| 22:26.11 | SkramX | I mostly do work with virtual and hosted asterisk :( |
| 22:26.23 | Trazz | i never see it register |
| 22:26.31 | Trazz | i have it configured to register |
| 22:26.45 | Trazz | 2004/2004 (Unspecified) D N 0 Unmonitored |
| 22:26.51 | Trazz | thats the status right now |
| 22:27.38 | harryvv | sucks that iax.cc blocks 1800 numbers outbound. some companies that are in the states advertise there 1800 number and nothing else. So need to edit my asterisk to force it to use whosale voip instead of zap |
| 22:27.57 | harryvv | those 1800 numbers dont work up here in canada. |
| 22:29.06 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:29.36 | *** join/#asterisk fugitivo (n=ajf@201.255.177.145) |
| 22:29.52 | fugitivo | any way to change automon filename? |
| 22:31.50 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:34.08 | Hmmhesays | got one install coming up and they have a "lignup" softswitch |
| 22:34.11 | Hmmhesays | as they called it |
| 22:34.41 | harryvv | What is that? |
| 22:35.04 | cyburdine | hey gang... I have an agi question should someone have time |
| 22:35.30 | Hmmhesays | can't answer if you don't ask |
| 22:35.41 | cyburdine | does anyone know how to pass information between AGI and a program being called by EXEC |
| 22:36.05 | cyburdine | I'm calling EXEC myprog |
| 22:36.44 | cyburdine | and it is then handeling the channel... and returning some text but I don't have access to that text |
| 22:36.44 | *** join/#asterisk ldav15 (n=ldavis@208.2.145.2) |
| 22:36.57 | cyburdine | hope that makes sense |
| 22:37.16 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 22:37.18 | fugitivo | cyburdine: just like any other program... |
| 22:38.59 | cyburdine | how so? |
| 22:39.53 | saitech | could you exec the program inside you agi script, and get the output from the exec in your script language? |
| 22:39.59 | *** join/#asterisk effape_ (n=nick@81.5.150.54) |
| 22:40.25 | cyburdine | that's what we are doing, but I don't know how to capture that text? throw to variable? |
| 22:40.38 | cyburdine | currently is dumps to stdout |
| 22:40.52 | saitech | if you are using php for instance, you can get stdout from a exec into a array ! |
| 22:40.53 | cyburdine | and shows up in console so I know my_prog works.. |
| 22:41.03 | fugitivo | for example |
| 22:41.12 | fugitivo | $foo = `date`; |
| 22:41.16 | fugitivo | that's an option for perl |
| 22:41.19 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:41.22 | fugitivo | but i don't know what language you're using |
| 22:41.33 | effape_ | hey guys me again :) |
| 22:41.40 | harryvv | hi |
| 22:41.42 | cyburdine | well the my_prog is c and I'm using perl for the AGI |
| 22:41.47 | cyburdine | script |
| 22:41.51 | effape_ | more problems as normal ;) |
| 22:41.58 | saitech | exec("myprog.sh", "arrOutput"); |
| 22:42.03 | saitech | use that in PHP |
| 22:42.08 | fugitivo | cyburdine: $var = `cprogram`; |
| 22:42.10 | saitech | dont know what to type in perl |
| 22:42.28 | fugitivo | cyburdine: $var will have the output of cprogram |
| 22:42.48 | saitech | fugitivo that will only work if its only one line of output, if its more than one it doesnt work. (in php) |
| 22:43.05 | effape_ | I've got the sip on incomming call doing what it should now. However the only problem i have now is if i call the zap interface then put the phone down after the sip has started ringing it keeps ringing even though the zap is down |
| 22:43.26 | *** join/#asterisk jerlique (n=jerlique@ppp103-178.static.internode.on.net) |
| 22:43.26 | cyburdine | hmmm except that we are doing processing on the audio... much like the record app does |
| 22:43.42 | fugitivo | saitech: perl will get all the lines |
| 22:44.04 | fugitivo | are you sure php don't? |
| 22:44.07 | cyburdine | so cprogram is taking the channel data as input and then dumping text |
| 22:44.28 | cyburdine | i.e. voice recognition |
| 22:44.47 | fugitivo | cyburdine: i don't know what cprogram does, i know that $var = `cprogram`; will put the output of cprogram in $var |
| 22:44.53 | saitech | fugitivo not quite. but the exec(program, array) is much easier to work along with, if you use $var = exec(program) you have to explode the lines. |
| 22:45.28 | cyburdine | saitech: so array will capture the output? |
| 22:45.43 | effape_ | shell_exec will give you a string |
| 22:45.45 | cyburdine | in your example exec[program, array] that is? |
| 22:45.50 | effape_ | in php |
| 22:45.55 | saitech | cyburdine: correct |
| 22:46.06 | effape_ | rather than an array |
| 22:46.11 | saitech | http://dk.php.net/function.exec |
| 22:46.14 | cyburdine | hrmm... very interesting... |
| 22:46.18 | saitech | you can se the documentation there |
| 22:46.23 | saitech | its a quite nice function |
| 22:46.36 | cyburdine | oh this is PHP's exec... not * exed |
| 22:46.41 | cyburdine | err exec |
| 22:46.43 | effape_ | or even backticks tbh in php they work fine |
| 22:46.58 | saitech | the return_var thing is also very nice |
| 22:47.21 | saitech | it could be nice, if * exec did the same :) |
| 22:47.21 | cyburdine | but I need a way to do this from * exec function |
| 22:47.34 | fugitivo | hey |
| 22:47.39 | fugitivo | perl is easier |
| 22:47.46 | cyburdine | pisser... I was hoping I had just missed something in perl |
| 22:47.47 | saitech | ohh. i would do it with an agi script then |
| 22:47.47 | fugitivo | @var = `cprogram`; |
| 22:47.51 | cyburdine | err. asterisk's docs |
| 22:47.52 | fugitivo | there, you have an array |
| 22:48.12 | saitech | ok then :) |
| 22:48.20 | cyburdine | fugitivo... will cprogram be recieving the channel audio data? |
| 22:48.26 | cyburdine | in your example? |
| 22:48.36 | fugitivo | that will receive stdout |
| 22:48.49 | fugitivo | if you send it to stdout... |
| 22:48.50 | cyburdine | right... by my cprog needs channel data in |
| 22:49.28 | fugitivo | maybe if you explain what are you trying to do... |
| 22:49.48 | jerlique | hi - can anyone explain the difference between using the register => namd:pass@x.x.x.x as opposed to listing the gateway as a 'friend' |
| 22:50.00 | cyburdine | we are answering the channel... then parsing the audio... (voice recognition) then dumping the result |
| 22:50.02 | cyburdine | in text |
| 22:50.35 | cyburdine | so we need a way to get both audio from the channel in to the cprog and text back out. |
| 22:50.35 | jbalcomb | Ah.. a very productive day. fixed so many issues my log files have become boring to watch. thanks everyone. |
| 22:51.41 | saitech | jerlique: if im not wrong. the "friend" thing, makes the call as an invite, if its an asterisk server, you have to use insecure=invite for it to authenticate the call. The register makes an register, so you can perform a call, without the insecure=invite. |
| 22:53.45 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 22:53.57 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 22:54.24 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.73) |
| 22:54.54 | jerlique | thanks saitech - will give it a try.... |
| 22:56.32 | harryvv | man my asterisk box just went unstable. Was getting a call via my 1800 voip number then wife decides to call. got lots of jitter on my 800 number and the call that came in from nevada showed a local 604 area code cid on the display. |
| 22:56.35 | Zodiacal | can someone dcc me from their /var/libs/asterisk/sounds/dir-intro-oper.gsm my asterisk release doesn't seem to include it... |
| 22:56.41 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 22:56.53 | Zodiacal | err |
| 22:56.56 | harryvv | now, the vm my wife left...is gone. |
| 22:56.57 | effape_ | anyone any ideas? |
| 22:57.06 | harryvv | as if asterisk deleted it. |
| 22:58.04 | saitech | jerlique: np ! |
| 23:01.05 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 23:01.50 | *** join/#asterisk Utah_Dav1 (n=boucha@0-2pool130-186.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:03.03 | *** join/#asterisk kc5cqm (n=kc5cqm@cpe-68-206-112-195.stx.res.rr.com) |
| 23:03.33 | kc5cqm | anyone here have experience with those cheapie "digium XP100s" (not really made by digium) |
| 23:03.41 | fugitivo | yes |
| 23:03.43 | fugitivo | they suck |
| 23:04.07 | kc5cqm | howso? I guess at $10 I'm not out too much |
| 23:04.23 | kc5cqm | but I'm wanting to tie my packet8 into my asterisk box now |
| 23:04.51 | kc5cqm | seems that's about the only way to do it that packet8 won't gripe about |
| 23:05.10 | kc5cqm | did you ever get yours working? I've got a shared IRQ problem |
| 23:05.20 | IMG-SD | I'm currently debugging a deadlock in Asterisk, and I'm stuck at about 75% following debugging instructions... can anyone explain to me how to do this step: 8) Check who has the lock |
| 23:05.20 | IMG-SD | Now if we have properly turned on thread debugging we are going to be able to see into include/asterisk/lock.f ast_mutex_t struct which looks like this |
| 23:05.20 | IMG-SD | <PROTECTED> |
| 23:05.21 | IMG-SD | <PROTECTED> |
| 23:05.21 | IMG-SD | <PROTECTED> |
| 23:05.23 | IMG-SD | <PROTECTED> |
| 23:05.25 | IMG-SD | <PROTECTED> |
| 23:05.27 | IMG-SD | <PROTECTED> |
| 23:05.29 | IMG-SD | <PROTECTED> |
| 23:05.31 | IMG-SD | so now that we have our lock we can see who has it & what we are waiting on, .. do the following bt cmds |
| 23:05.33 | IMG-SD | <PROTECTED> |
| 23:05.35 | IMG-SD | <PROTECTED> |
| 23:05.37 | IMG-SD | <PROTECTED> |
| 23:05.39 | IMG-SD | <PROTECTED> |
| 23:05.41 | IMG-SD | <PROTECTED> |
| 23:05.43 | IMG-SD | <PROTECTED> |
| 23:05.45 | IMG-SD | This is the guility code that is holding the lock that we want to look at. |
| 23:07.05 | kc5cqm | for setting up 4 pstn connections, is there really any advantage whatsoever with using ISDN? |
| 23:07.22 | effape_ | i imagine it's better that trying to figure out pstn ;) |
| 23:07.36 | effape_ | and isdn cards are like 5 quid on ebay ;) |
| 23:08.52 | r_evolution | just because it's been a minute... ever feel like this : http://www.syslog.com/~jwilson/pics-i-like/21_podborka_4.jpg |
| 23:09.07 | kc5cqm | isdn service is more pricey than pstn here in the US though.. |
| 23:09.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:09.58 | r_evolution | hey kc... why is packet8 having such a fit about you tying your box into *? |
| 23:10.13 | effape_ | yeah working! |
| 23:10.25 | r_evolution | effape... you've gotten it to detect a hangup as well? |
| 23:10.26 | effape_ | lol evo btw |
| 23:10.42 | effape_ | yeah. Just seems to take 3-4 secs |
| 23:11.01 | r_evolution | yeah... that's pretty much what I've got it doing here... listening for silence and a repeated busy signal |
| 23:11.11 | effape_ | but i imagine that could be because i'm on the xlite over a sat connection to the asterisk box. |
| 23:11.19 | r_evolution | over a sat conn? O_O |
| 23:11.21 | effape_ | it's got a 1sec lag |
| 23:11.40 | effape_ | it's because the guy i'm setting this up for is in the middle of nowhere |
| 23:11.47 | effape_ | can't get adsl or anything |
| 23:11.58 | r_evolution | eh. I'm working with a company who seems to be slow as hell about dropping the 10$ to get 729 licensed... it's getting to the point where I'm like.. look... ILL PAY IT JUST STFU ALREADY |
| 23:12.00 | r_evolution | good deal :) |
| 23:12.00 | effape_ | so it's 2 meg sat but lag is a killer |
| 23:12.05 | r_evolution | im sure man |
| 23:12.12 | r_evolution | it's going like 20,000 miles off the planet |
| 23:12.13 | r_evolution | and coming back |
| 23:12.16 | effape_ | yeah |
| 23:12.21 | effape_ | MY VOICE IS argh |
| 23:12.24 | effape_ | foil hats |
| 23:12.28 | r_evolution | haha |
| 23:12.38 | r_evolution | :( |
| 23:12.44 | effape_ | heh |
| 23:12.57 | r_evolution | but it's good that you've gotten everything working correctly. |
| 23:13.23 | effape_ | yeah. Now i have the big problem of BT not being able to provide dring on the lines :( |
| 23:13.44 | r_evolution | I'm in the US and it's SUPPOSED to detect a hang-up based on polarity switch... but some for some reason... |
| 23:14.01 | r_evolution | what kind of phone is he using? |
| 23:14.08 | r_evolution | xlite? or is that just what you're using? |
| 23:14.20 | effape_ | no that's just me. He has 3 snom 360s |
| 23:14.31 | effape_ | and i'm putting in a cisco ata186 too |
| 23:14.37 | r_evolution | ahh... that's what we're testing with here |
| 23:14.42 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 23:14.42 | r_evolution | that and the polycom 301s |
| 23:14.51 | r_evolution | the sysadmin was the one who wanted the snoms ;x |
| 23:15.07 | effape_ | but he has 3 lines (2 voice and 1 fax) but i'm needing to the one voice to one phone and the other to the rest. |
| 23:15.12 | *** part/#asterisk IMG-SD (n=IMG-SD@64.5.206.131) |
| 23:15.28 | effape_ | aha cool. It's my first attept at an install so i just looked at the snoms |
| 23:15.31 | r_evolution | k so what's the problem? |
| 23:15.37 | r_evolution | don't feel bad... mine too ;x |
| 23:15.42 | effape_ | i'm using AMP |
| 23:15.42 | effape_ | :) |
| 23:15.43 | r_evolution | i've just been reading anything i can touch |
| 23:15.46 | effape_ | hehe |
| 23:15.55 | r_evolution | it's like *touchreadtouchreadtouchread* |
| 23:15.57 | *** join/#asterisk dweebus_compleat (n=root@71.224.35.58) |
| 23:16.02 | effape_ | i've subscribed to safari.oreilly - that asterisk book is on there |
| 23:16.18 | r_evolution | well you can actually DL the pdf of the oreilly * for free man :) |
| 23:16.25 | effape_ | i don't know how to make it do that. The only way i could figure that it would be able to detect which line it was was dring |
| 23:16.29 | r_evolution | i bought it months before I DLd but if you want I'd be happy to DCC it to you |
| 23:16.40 | effape_ | yeah found that out after. I use it for all my other stuff to ;) |
| 23:16.44 | r_evolution | well... you could just sent it to fwd a specific number to a specific phone |
| 23:16.47 | effape_ | pro php programmer by trade ;) |
| 23:16.56 | *** part/#asterisk UlbabraB_ (n=salama@host241-43.pool8172.interbusiness.it) |
| 23:17.00 | r_evolution | ahh... i dont know wtf i am by trade |
| 23:17.05 | effape_ | lol |
| 23:17.06 | r_evolution | im a "hey you go do this" by trade |
| 23:17.14 | effape_ | would that by by zap interface? |
| 23:17.18 | r_evolution | graphics? web-sites? servers? networks? etc etc |
| 23:17.19 | effape_ | heh |
| 23:17.21 | Zodiacal | is there an easy way to disable the directory? i.e. # |
| 23:17.26 | r_evolution | nah you'd do it in extensions.conf |
| 23:17.39 | r_evolution | you've got the zap interface right? so how many fxo ports? |
| 23:17.44 | r_evolution | i'm assuming two |
| 23:17.56 | effape_ | 3 |
| 23:17.59 | effape_ | yeah |
| 23:17.59 | r_evolution | so have zap/1 go to phone 1... and put zap/2 into the queue |
| 23:18.04 | r_evolution | yeah but one is for faxing |
| 23:18.07 | effape_ | yep |
| 23:18.08 | r_evolution | TDM400P? |
| 23:18.10 | effape_ | yep |
| 23:18.16 | r_evolution | why use a distinctive ring? |
| 23:18.34 | effape_ | the only way i could figure it without messing up amp |
| 23:19.05 | r_evolution | i've never used amp... i've been all CLI since i started messing with it... i'm just starting to look around anything else |
| 23:19.10 | effape_ | extensions.conf being 781 lines of bloat ;) |
| 23:19.15 | r_evolution | =-o |
| 23:19.26 | effape_ | yeah |
| 23:19.52 | r_evolution | well |
| 23:20.08 | r_evolution | are both coming into an IVR and then going out from there? or is one ringing directly to an extension? |
| 23:20.44 | effape_ | during the both will goto different extensions but in the evening both will goto the same |
| 23:21.16 | effape_ | during [the day] ;) |
| 23:21.37 | r_evolution | i figured :-D |
| 23:21.41 | r_evolution | well why not set up an IVR? |
| 23:21.41 | effape_ | heh |
| 23:21.48 | r_evolution | have both come in the IVR |
| 23:22.00 | r_evolution | for "such and such" press 1.... for "so and so" press two |
| 23:22.06 | r_evolution | that way people can call either line for either person |
| 23:22.25 | r_evolution | then you set it up so after a specific time... both go to the one person |
| 23:22.34 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.130) |
| 23:23.05 | effape_ | yeah it's more that they have the home and the office in the same place. But they have a home number which should always ring the home phone but the office line ring to the office or voicemail |
| 23:23.21 | effape_ | but the two lines can be use for outgoing from any phone |
| 23:23.33 | r_evolution | ohhhh |
| 23:23.35 | effape_ | ivr not really suitable for home ;) |
| 23:23.37 | Seldon1975 | why does my Polycom 501 waste a whole SpeedDial hardkey by assigning it to my own extension? Can I replace this? |
| 23:23.45 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 23:24.24 | r_evolution | what about if you set the context to be different for each zap channel in zapata.conf? |
| 23:24.37 | r_evolution | remember im fairly new at this as well... so i'm suggesting things i might try... |
| 23:25.09 | effape_ | yeah absolutly... i'm open to anything :) |
| 23:25.25 | r_evolution | mmmk... well go into the zapata.conf where you first set up the zap channels... |
| 23:25.37 | r_evolution | and in there you probably have like "context=incoming" or whatever... |
| 23:25.45 | effape_ | yep |
| 23:25.48 | r_evolution | so... divide it into different channels... |
| 23:25.56 | r_evolution | Channel => 1 and channel => 2 |
| 23:26.02 | r_evolution | different groups etc |
| 23:26.19 | r_evolution | and point each channel at a different context |
| 23:26.35 | r_evolution | then you can set it to at a certain time transparently pass to the one extension |
| 23:26.48 | r_evolution | not sure if that would work... but i think it'd at the very least be worth a shot... |
| 23:26.54 | effape_ | using context-to= ? |
| 23:27.01 | r_evolution | nono |
| 23:27.05 | r_evolution | context => |
| 23:27.14 | r_evolution | context => 'contextnamehere' |
| 23:27.25 | effape_ | aha cool. I'll give that a shot |
| 23:27.28 | effape_ | cheers |
| 23:27.42 | r_evolution | yeah let me know how it goes im curious myself |
| 23:27.49 | effape_ | yeah defo |
| 23:27.49 | fndude | Is there a secret to using the "s" extension? Do I have to include it in my register=>? or can I dial that extension somehow when an outside call comes in? |
| 23:28.07 | r_evolution | s is the extension used when no other extension is defined |
| 23:28.19 | r_evolution | like... if a call is just dumped into a context... it goes to S |
| 23:28.21 | r_evolution | like start :) |
| 23:28.49 | r_evolution | ok im going to setup some shiot... aye eff kay. |
| 23:28.53 | r_evolution | or at least this set =\ |
| 23:29.14 | gaz00 | anyone know of an front-door style intercom that I can use that would connect to Asterisk? |
| 23:29.33 | Trazz | Jan 17 17:41:19 NOTICE[31195]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:29.35 | *** join/#asterisk _cleric_ (n=dacleric@p54828DBF.dip0.t-ipconnect.de) |
| 23:29.35 | Trazz | ? |
| 23:29.39 | tainted_ | any IAX experts here? |
| 23:30.40 | tainted_ | if Box A passes a call to Box B, and Box B is on a totally different network, does the call re-route to Box B or still goes through Box A for the duration of the call? |
| 23:31.01 | badboyz | do you have to save an AA .wav file in a special format for * to play it? |
| 23:31.23 | Trazz | tainted, is the call coming in via t1 or pri , isdn ? |
| 23:31.44 | tainted_ | Trazz voip call |
| 23:31.57 | *** join/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 23:32.04 | r_evolution | hey trazz... would that imply that you're possibly trying to pass it to a SIP channel that hasn't been setup? |
| 23:32.09 | r_evolution | wouldn't* |
| 23:32.26 | r_evolution | like if you had ext 2203 setup in sip.conf... and you were trying to pass it to 2230 |
| 23:32.28 | Trazz | r_evolution, well the extension exists |
| 23:32.44 | iccomputing | Anyone have any idea why I keep getting 401 Unauthorized in my Sip Debug? |
| 23:32.52 | Trazz | i agree but the extension does exist |
| 23:32.52 | r_evolution | have you called from the extension you're trying to call?> |
| 23:33.08 | r_evolution | hmm. |
| 23:33.19 | Trazz | the extesion is available |
| 23:33.19 | tainted_ | Trazz so does Box A stay in the picture? |
| 23:33.34 | r_evolution | are you telling * to qualify it? |
| 23:33.35 | Trazz | trained, i believe it has to stay in the picture |
| 23:33.50 | Trazz | r_evolution , how ? |
| 23:33.51 | tainted_ | hmm |
| 23:34.03 | iccomputing | Anyone have experience with AASTRA/SAYSON 480i configs? |
| 23:34.21 | r_evolution | in sip.conf |
| 23:34.24 | r_evolution | qualify=yes |
| 23:34.40 | iccomputing | me r_evolution? |
| 23:35.12 | r_evolution | no. Trazz |
| 23:35.22 | iccomputing | ohh...ok.. |
| 23:35.26 | fndude | My sip channel thats registered does not seem to like being tossed into a context without defining an extension that matches my DID, is this normal behavior? |
| 23:35.32 | Trazz | call comes in over voip and hits * box. when i transfer i get that.. |
| 23:35.44 | Trazz | the * box and the extension are on same network |
| 23:35.45 | Trazz | no nat |
| 23:35.52 | Trazz | nat only to broadvoice |
| 23:36.08 | Trazz | i can try qualify too |
| 23:36.27 | r_evolution | are you trying to setup your box to register with broadvoice? and having problems with that? |
| 23:36.40 | Trazz | no the box is registered and working great |
| 23:36.44 | Trazz | i dial in to basic ivr |
| 23:36.52 | Trazz | hit 5 and it goes to unavilable |
| 23:36.52 | r_evolution | if I'm not mistaken... there was something more special you had to do in order to make broadvoice work properly... |
| 23:37.01 | r_evolution | pastebin your extensions.conf |
| 23:37.06 | Trazz | ok 1 sec |
| 23:37.06 | r_evolution | or at least the ivr place... |
| 23:37.07 | r_evolution | plz |
| 23:37.10 | snova | for those which just joined.. anyone know of inexpensive VoIP to GSM gateway? |
| 23:37.10 | r_evolution | thx :) |
| 23:37.56 | r_evolution | and throw rocks at me when you do that... b/c i am working on something here... soo... |
| 23:38.30 | Trazz | http://pastebin.com/510711 |
| 23:38.58 | snova | anyhow have any positive experience with SIPPhone inbound using their virtual number service? |
| 23:39.45 | Trazz | r_evolution here is the other part |
| 23:39.46 | Trazz | http://pastebin.com/510714 |
| 23:40.23 | Trazz | so broadvoice is in the from-dip context |
| 23:40.42 | iccomputing | anyone have some time to help teach me some things?? |
| 23:41.16 | snova | I got time to teach you some things but they probably would be of any interest to you |
| 23:42.19 | iq | Hi, Anyone working on CSTA enablement of Asterisk? |
| 23:42.20 | snova | how about, how to drink milk with a straw using your nose? |
| 23:42.26 | *** part/#asterisk Utah_Dav1 (n=boucha@0-2pool130-186.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:42.34 | r_evolution | agh snova... ouch! |
| 23:42.41 | Netgeeks | CSTA? |
| 23:42.48 | r_evolution | hey Trazz... when you dial ext 2000 without using the IVR... does it go through? |
| 23:43.09 | Trazz | it says i am on the phone right now |
| 23:43.10 | iq | Netgeeks, Computer-supported telecommunications applications |
| 23:43.13 | Trazz | when i dial from my side |
| 23:43.13 | tainted_ | if i have a user [ata] on asterisk box [asteriskA], what's the dialstring to call [ata] from [asteriskB] |
| 23:43.18 | Trazz | i can dial from console |
| 23:43.45 | Netgeeks | iq: ah, not familair with the acronym... can you give an example? |
| 23:44.12 | Netgeeks | or is CSTA a particular spec? |
| 23:44.13 | iq | Netgeeks, CTI? |
| 23:44.33 | Trazz | goes to vm |
| 23:44.35 | iq | Netgeeks, http://en.wikipedia.org/wiki/CSTA |
| 23:44.41 | r_evolution | ok... so take all extra options off... voicemail and everything just comment them out... |
| 23:44.49 | snova | GSM Gateway? anyone, anyone?.... crickets crickets crickets.... |
| 23:45.21 | r_evolution | so it goes straight to voicemail... perhaps the phone is not registering... try putting qualify=yes in the section in your sip.conf defining SIP 2000 |
| 23:45.22 | Netgeeks | CTI I do recognize |
| 23:45.27 | Trazz | Jan 17 17:57:27 NOTICE[31382]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:45.28 | Trazz | <PROTECTED> |
| 23:45.28 | snova | someone tell me to go take make my GSM Gateway questions and go stick it where the sun doesn't shine |
| 23:45.35 | Trazz | ok |
| 23:45.51 | r_evolution | because it really looks like the phone is not registering... |
| 23:46.35 | r_evolution | b/c your dial-plan looks straight to me... well other than the fact that you could be using a macro for a lot of the typing you're doing... |
| 23:47.02 | r_evolution | and i think your unavailable in the dialplan should be exten => 2001,2,Voicemail(u2001@VOICEMAILSECTIONNAME) |
| 23:47.04 | Netgeeks | iq: I don't see any evidence of anyone publicly working on CTSA for Asterisk, but I would bet someone is |
| 23:47.21 | Trazz | i can try that |
| 23:47.39 | iq | Netgeeks, yes I know some people who are working. I was looking for someone in this channel. Thanks anyways :) |
| 23:47.52 | lesouvage | iccomputing: what's the question. |
| 23:48.22 | lesouvage | iccomputing: you shouldn't ask to ask |
| 23:49.24 | iccomputing | hey lesouvage...sorry I got a phone call!! |
| 23:49.52 | tainted_ | anyone know? |
| 23:49.54 | tainted_ | if i have a user [ata] on asterisk box [asteriskA], what's the dialstring to call [ata] from [asteriskB] |
| 23:51.20 | iccomputing | lesouvage: I need to learn how to correctly build a sip config for an AASTRA 480i....I got a good handle on the aastra.cfg and [mac].cfg (for the phone itself)...but inside asterisk, i am having problems... |
| 23:52.00 | lesouvage | iccomputing: what flavor of asterisk are you running? |
| 23:52.07 | *** join/#asterisk |omni| (n=rob@net98.limelyte.net) |
| 23:52.11 | iccomputing | asterisk @ home... |
| 23:52.26 | iccomputing | i couldnt figure out how to make regular asterisk 1.2 work |
| 23:52.52 | lesouvage | icomputing: go to /etc/asterisk/extensions.conf |
| 23:53.04 | iccomputing | i have been digging deeper and deeper into this...I have a lot of experience with hosted PBX's like broadsoft...but I like the control that asterisk gives me.. |
| 23:53.09 | iccomputing | ok |
| 23:53.10 | iccomputing | there |
| 23:53.59 | lesouvage | icomputing: there should be a line like # include sip-phones.conf or something like that (I'm not familiar with A@H but I am with asterisk) |
| 23:54.38 | Netgeeks | I've written a broadsoft competetive product with asterisk... |
| 23:55.01 | Netgeeks | basically allowing anyone to become a broadsoft |
| 23:55.12 | iccomputing | lesouvage: #include extensions_additional.conf |
| 23:55.35 | iccomputing | Netgeeks..i can see why that would be good |
| 23:56.09 | lesouvage | iccomputing: I opened a private channel so we don't fload the channel, canyou see it. |
| 23:56.18 | iccomputing | ahh yes |
| 23:57.10 | litage | why do so many asterisk admins use CentOS? |
| 23:57.24 | gaz00 | why not? |
| 23:57.40 | Netgeeks | kinda asking why do so many people drive fords? |
| 23:58.23 | litage | not really. so many people drive fords because they're [generally] cost-effective, reliable and economical while providing a decent array of features |
| 23:58.33 | effape_ | mini mini mini :D |
| 23:58.36 | gaz00 | Except one costs a little more than $10, and the other is open source. |
| 23:58.43 | litage | heh |
| 23:58.46 | Netgeeks | but it's the same thing |
| 23:59.01 | litage | gaz00: i'm curious as to why so many people choose centos rather than fedora, debian, etc |
| 23:59.02 | Netgeeks | people use CentOS / drive fords because they have a personal reason to do so |
| 23:59.19 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 23:59.20 | gaz00 | A@H has a lot to do with it, I think. |
| 23:59.27 | *** join/#asterisk secure75 (n=mic@dslb-084-057-039-156.pools.arcor-ip.net) |
| 23:59.30 | Netgeeks | asterisk doesn't perform any better on CentOS than on Fedora or debian, etc. |
| 23:59.33 | gaz00 | I first installed it on SuSE, didn't like it. |
| 23:59.39 | gaz00 | same with Debian. |
| 23:59.42 | litage | gaz00: ah, that definitely contributes to centos's use |
| 23:59.56 | litage | gaz00: why did you not like * on debian? |
| 23:59.57 | gaz00 | CentOS is under A@H, easiest install EVER |
