| 00:02.50 | JunK-Y | antitrust on tv, what a movie! |
| 00:02.58 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:02.58 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:03.09 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - Chicago |
| 00:03.58 | salviadud | could somone give me a call? 651692 on fwd |
| 00:04.04 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:04.07 | salviadud | just configured stun |
| 00:04.13 | *** join/#asterisk chipach (~chipach@gatekeeper.athenahealth.com) |
| 00:04.20 | salviadud | and the automated thing can't call me, for some reason... |
| 00:05.28 | *** join/#asterisk Ryan_Finn (~rfinnesey@cpe-69-203-74-122.nyc.res.rr.com) |
| 00:05.39 | shmaltz | intersting: |
| 00:05.40 | shmaltz | http://news.yahoo.com/s/cmp/20050504/tc_cmp/162101118/nc:1293;_ylt=AvzHQelNdE6OMD72z9zRyv_67rEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl |
| 00:07.55 | Ryan_Finn | Hi All |
| 00:08.22 | Ryan_Finn | I am new to Asterisk some please let me know if i am off base |
| 00:09.01 | shmaltz | Ryan_Finn, go ahead |
| 00:09.48 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 00:09.50 | salviadud | ryan, the difference between fxo and fxs |
| 00:10.08 | Ryan_Finn | who can I do to about getting numbers in Singapore? |
| 00:10.12 | *** join/#asterisk chipach (~chipach@gatekeeper.athenahealth.com) |
| 00:10.43 | Micc | In the current iaxclient cvs-head are there any examples for windows that build besides the testcall? |
| 00:11.11 | Micc | It seems the samples are all out of date. |
| 00:13.12 | Ryan_Finn | I have looked on the web and I see most people offering UK and US numbers but I do not see anything for Singapore |
| 00:14.41 | MikeJ[Laptop] | you are looking for singapore did's? |
| 00:14.41 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 00:14.48 | *** join/#asterisk mwgbc (~mwallace@adsl-68-126-232-10.dsl.pltn13.pacbell.net) |
| 00:14.54 | Ryan_Finn | Yes I am |
| 00:14.58 | MikeJ[Laptop] | one sec |
| 00:15.12 | *** join/#asterisk Atacomm (~dan@69.54.45.98) |
| 00:15.17 | salviadud | yay! |
| 00:15.18 | Ryan_Finn | Thank you |
| 00:15.19 | salviadud | i got it working |
| 00:15.37 | MikeJ[Laptop] | maybe, I know somone who might have some |
| 00:15.48 | mwgbc | Will asterisk's Voicemail system read wav files that don't have header info? |
| 00:16.00 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 00:16.02 | Ryan_Finn | Ok that would be very helpful Thank you |
| 00:16.17 | Ryan_Finn | Would you like my contact info? |
| 00:16.23 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 00:17.21 | MikeJ[Laptop] | sorry, no go |
| 00:18.19 | Ryan_Finn | that is ok thank you for your time |
| 00:18.36 | mwgbc | I have a program that generates wav files and sticks them in a specific directory I want to use asterisk to call someone and read the file to them. Where do I start? |
| 00:19.27 | marlowe | mwgbc : Start on voip-info.org.. Try the Dial cmd - It's a built in function. |
| 00:19.38 | Ryan_Finn | MikeJ: Where is the best place to look for DID's? |
| 00:20.00 | mwgbc | marlowe : Yes, I understand that part, sorry, my question wasn't clear... |
| 00:20.04 | marlowe | Ryan_Finn: There is no "best place" - Look on voip-info or the asterisk biz mailing list. Look around, take a pick. |
| 00:20.15 | *** join/#asterisk IPmonger (~ipmonger@68.84.208.206) |
| 00:20.49 | mwgbc | marlowe : I was wondering if asterisk would use a wav file with no header info in it. |
| 00:21.00 | *** part/#asterisk IPmonger (~ipmonger@68.84.208.206) |
| 00:21.07 | marlowe | Asterisk plays nice with gsm - Why not just use gsm? |
| 00:21.18 | JunK-Y | ~seen steveu |
| 00:21.21 | jbot | i haven't seen 'steveu', JunK-Y |
| 00:21.30 | JunK-Y | ~seen coppice |
| 00:21.31 | jbot | coppice <~chatzilla@60.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 10h 31m 59s ago, saying: 'ManxPower: most clueless on the internet, maybe :-)'. |
| 00:21.46 | salviadud | is there a thirdy party prog i can use to make GSM files??? |
| 00:21.50 | salviadud | in linux? |
| 00:22.12 | Atacomm | ~seen denon |
| 00:22.14 | jbot | denon is currently on #asterisk (15h 53m 25s). Has said a total of 1 messages. Is idling for 8h 13m 51s |
| 00:22.21 | mwgbc | marlowe : The proprietary program records wavs without header info they are 16bit 8000 Mono files |
| 00:23.04 | JunK-Y | marlowe: sometime we need to use other thing then gsm |
| 00:23.37 | salviadud | what good recording audio prog on linux or unix? |
| 00:24.16 | cyt0plas | I think I've got Asterisk talking properly to my incoming T1 PRI, how would I test that? |
| 00:24.25 | mwgbc | Unfortunatly the program is Win based and I will have to use the win version of Asterisk for this project |
| 00:25.09 | mwgbc | marlowe : Any ideas about how to handle the wav files? |
| 00:25.34 | JunK-Y | from gsm ya can generate a wav easily. |
| 00:26.32 | shepherd | salviadud: sox |
| 00:27.31 | mwgbc | Junk-Y : I have a program that already generates wav files (it is a predictive dialer) i need to have asterisk read those over a connection it establishes (to my cell) |
| 00:27.34 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 00:28.27 | JunK-Y | Playback? |
| 00:28.44 | JunK-Y | im not dealing with wav anyways, just with gsm,ulaw |
| 00:31.19 | mwgbc | I guess I'll figure somthing else out |
| 00:31.28 | mwgbc | Thanks for your help |
| 00:31.37 | TechDawg | I'm trying to setup an exten to dial the PSTN network. Can someone provide me an example? |
| 00:32.30 | TechDawg | Well, more specific, can someone tell me how I can test an outbound from one * to another * with an outbound using CLI? |
| 00:32.54 | TechDawg | They are registered using IAX2 |
| 00:33.22 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:34.27 | mwgbc | VV#47 VV3 #4V3 #3P\3 !5 4 |=4!|_(_)P\3 7() [()^^(_)|\|![473 |
| 00:34.58 | *** part/#asterisk mwgbc (~mwallace@adsl-68-126-232-10.dsl.pltn13.pacbell.net) |
| 00:35.16 | TechDawg | Well that was rather profound. |
| 00:35.39 | shepherd | mwgbc: try audacity |
| 00:36.09 | TechDawg | Alrighty then, didn't make any sense to me. |
| 00:36.39 | shepherd | ooops.. too late |
| 00:36.40 | shepherd | haha |
| 00:36.46 | TechDawg | Just a tad. |
| 00:37.09 | drbrown | is there a way to alter the dial plan according to time of day? |
| 00:37.17 | TechDawg | yes |
| 00:37.28 | drbrown | how is that accomplished? |
| 00:37.49 | TechDawg | Uhm, depends on how you want to do it. |
| 00:38.14 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 00:38.16 | TechDawg | There are several include examples on the net but I ran across a dialplan that didn't use the includes. |
| 00:38.18 | drbrown | I want my dial plan to bring up ivr prompts after work hours |
| 00:38.25 | kFuQ | exten => s,1,GotoIfTime(8:30-22:00|sun-sat|*|*?other_part_of_dialplan,s,1) |
| 00:38.42 | kFuQ | similar to that |
| 00:38.45 | TechDawg | There, that's the better way. |
| 00:38.45 | drbrown | great |
| 00:39.03 | drbrown | gotoiftime |
| 00:39.08 | drbrown | that's great |
| 00:39.14 | kFuQ | lookup GotoIfTime on voip-info |
| 00:39.16 | drbrown | thanks |
| 00:39.19 | kFuQ | np\ |
| 00:39.22 | drbrown | I will |
| 00:39.29 | TechDawg | I just wish I were at that point of my setup. |
| 00:39.37 | drbrown | need help |
| 00:39.39 | drbrown | ?? |
| 00:40.06 | cyt0plas | Anyone willing to help with with an asterisk T1 setup? |
| 00:40.07 | drbrown | I am about to rewrite my dial plan using macros, but have not started |
| 00:40.22 | TechDawg | I'm trying to setup a dialplan that is initiated by one * and transfered to another to dial out the PSTN drbrown |
| 00:41.35 | drbrown | TechDawg: all you have to do is put a goto on the second asterisk initiated from the extension of the first |
| 00:41.40 | drbrown | that should be real easy |
| 00:41.57 | drbrown | similar to answering from an 800 number from asterlink |
| 00:41.59 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
| 00:43.16 | drbrown | TechDawg: example: exten => 800blahblah,1,Goto(asterisklink,s,Context) |
| 00:43.36 | TechDawg | hmmm |
| 00:43.39 | drbrown | should be something like that |
| 00:43.40 | TechDawg | s? |
| 00:43.47 | drbrown | start extension |
| 00:44.12 | TechDawg | Ah, I think I see |
| 00:44.16 | drbrown | you can use a macro, or you can use another (brain fart) 1,2,3,4 |
| 00:44.32 | drbrown | s as in start |
| 00:44.40 | drbrown | like I said brain fart |
| 00:44.54 | drbrown | priority |
| 00:44.57 | drbrown | there |
| 00:45.11 | salviadud | drbrown, why brain fart? |
| 00:45.34 | drbrown | salivadud: couldn't think of what to call the s |
| 00:45.44 | salviadud | ah, the start thingy |
| 00:45.49 | drbrown | yes |
| 00:45.55 | salviadud | well, you could call it a lot things, i get ya |
| 00:45.58 | TechDawg | So like box one exten=_9.,1,Dial(IAX2/user:pass@boxtwo/s/outbound${EXTEN}) |
| 00:46.13 | drbrown | yes |
| 00:46.19 | *** join/#asterisk Rick_Hunter (~rhunter@02-041.008.popsite.net) |
| 00:46.44 | drbrown | wouldn't use 9, setup an entire outbound solution so the user doesn't have to dial the 9 |
| 00:47.03 | drbrown | I will send you a pastebin link |
| 00:47.06 | TechDawg | Oh and to strip the 9 change ${EXTEN} to ${EXTEN:1} eh? |
| 00:47.18 | drbrown | Wild cards |
| 00:47.37 | drbrown | in my dial plan it picks an extension based on the number dialed |
| 00:47.48 | drbrown | if it is local it goes out the first two |
| 00:47.51 | TechDawg | The idea is to be able to dial 9 to connect to a server in one NXX and dial 8 to connect to a server in another NXX |
| 00:48.16 | drbrown | why? |
| 00:48.21 | drbrown | out of curiosity |
| 00:48.24 | Nugget | why require the prefixes if the NXXs are distinct? |
| 00:48.30 | drbrown | automate it based on the number dialed |
| 00:48.39 | drbrown | exactly |
| 00:48.39 | Nugget | you can do that without silly prefix dialing |
| 00:48.41 | TechDawg | They aren't distinct |
| 00:49.03 | drbrown | can you give an example? |
| 00:49.16 | drbrown | ignorepat => 9 |
| 00:49.32 | TechDawg | i.e. 830-672 is different than 830-249 and we will have a box in each location connecting to a box in the 830-372. |
| 00:49.40 | *** join/#asterisk bjohnson (~bjohnson@ip206-172.dsl.istop.com) |
| 00:49.59 | drbrown | you can automate it |
| 00:50.03 | TechDawg | The idea is to use local PSTN lines we have to make calls to each location. |
| 00:50.15 | drbrown | easy |
| 00:50.25 | Ryan_Finn | Where would be the best place to find a * consultant based in New York? |
| 00:50.32 | TechDawg | So I've been told, and that's what I'm trying to get setup. |
| 00:50.47 | drbrown | I can show you |
| 00:50.55 | drbrown | hold on |
| 00:50.57 | Nugget | so you are routing based on LATA within an area code? still easy to do without prefixes. |
| 00:51.25 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
| 00:51.29 | TechDawg | Yes Nugget, I've been told this but I'm still trying to get the routing working in the same NXX |
| 00:51.38 | TechDawg | I'm really new to * |
| 00:51.48 | drbrown | me as well |
| 00:51.51 | drbrown | 2 weeks |
| 00:52.10 | TechDawg | Dang, I've been muddling over it for over a year. |
| 00:52.10 | Weezey | _NXXNXXXXXX,1,Dial(SIP/somecontext/${EXTEN}) |
| 00:53.15 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 00:53.29 | cyt0plas | Alright, I'm desperate - I've got $15 paypal, if someone is willing (and able) to get me started :P |
| 00:53.41 | TechDawg | Close but not quite Weezey. I want to pick up a phone on the FXS interface and call out to _NXXNXXXXX and have it routed through IAX2 to another box which will place the call on the FXO. |
| 00:54.04 | TechDawg | No SIP devices |
| 00:54.07 | drbrown | or exten => _NXXXXXX,2,Dial(IAX2/number@asteriskbox/${EXTEN}) |
| 00:54.35 | drbrown | 1234567 would dial just 1234567 |
| 00:54.43 | drbrown | and route it where you want |
| 00:55.11 | Weezey | Tech: you just need to set up IAX2 incoming on the other box in whatever context that accepts NXXNXXXXXX |
| 00:55.58 | Weezey | cyt0plas: where are you stuck? |
| 00:56.45 | TechDawg | Okay, that confirms it. Right now, I have a loop. * box one forwards to * box two which then tries to ring back to * box one. |
| 00:56.59 | Weezey | heh |
| 00:57.05 | TechDawg | I'll have to rethink my context and dialplans. |
| 00:57.35 | TechDawg | something about the default context that keeps causing the loop |
| 00:58.06 | Weezey | your IAX incoming should go do an outgoing context' |
| 00:58.53 | drbrown | don't use the context default, and break up the dial plan as much as possible |
| 00:59.02 | TechDawg | Right now, I have a default context that reads exten=s,1,Dial(IAX2/user:pass@boxone/24) |
| 00:59.04 | drbrown | be very specific with your contexts |
| 00:59.21 | Weezey | that should be a number not a s |
| 00:59.22 | drbrown | don't use default context |
| 00:59.29 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 00:59.47 | TechDawg | The problem that I have is if I don't have a default context, incoming calls on the FXO don't do anything. |
| 01:00.10 | Weezey | guh, the incoming should have it's own context too |
| 01:00.11 | drbrown | weezey is right you have no number on incomming |
| 01:00.22 | TechDawg | they try internal,s,1, then give up and try default |
| 01:01.01 | TechDawg | So I need to define an incoming context?? |
| 01:01.13 | drbrown | I would |
| 01:01.15 | Weezey | what kind of device is your FXO? |
| 01:01.23 | drbrown | it breaks up the dial plan further |
| 01:01.33 | TechDawg | Clone X100P |
| 01:01.34 | drbrown | are you sure the fxo is configured properly? |
| 01:01.49 | drbrown | ztcfg -v |
| 01:01.55 | Weezey | TechDawg: so in zapata.conf set context=incomingfxo |
| 01:02.14 | drbrown | exactly |
| 01:02.23 | Weezey | then in extensions.conf create [incomingfxo] with whatever you want to do with incoming calls in it.' |
| 01:02.42 | drbrown | then add includes for the other contexts it needs to access |
| 01:02.46 | Weezey | yeah |
| 01:02.54 | TechDawg | Okay, so what about outgoing? |
| 01:02.56 | Weezey | like include=>vomit |
| 01:03.09 | drbrown | exactly |
| 01:03.09 | Weezey | outgoing on the other box? |
| 01:03.13 | TechDawg | ztcfg shows two channels |
| 01:03.20 | drbrown | are they up? |
| 01:03.28 | drbrown | good |
| 01:03.31 | Weezey | X100P has one FXO and FXS? |
| 01:03.44 | TechDawg | Sorry wrong card |
| 01:03.52 | drbrown | is one the ztdummy? |
| 01:03.56 | TechDawg | They are the intel ambients |
| 01:03.58 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 01:04.03 | drbrown | how many? |
| 01:04.09 | TechDawg | two |
| 01:04.13 | Weezey | guh |
| 01:04.18 | Weezey | one is one channel one is the other |
| 01:04.29 | TechDawg | yes |
| 01:04.45 | Weezey | so what's the problem? |
| 01:04.46 | drbrown | I thought that the ambiant cards could only be used as fxo's |
| 01:04.52 | TechDawg | It's not hardware, it's my friggin dialplan and not getting it right. |
| 01:05.12 | drbrown | can you goto pastebin.ca and paste it? |
| 01:05.21 | TechDawg | The * box one has the FXS interface |
| 01:05.59 | drbrown | the extensions.conf |
| 01:06.05 | drbrown | or most of it |
| 01:06.14 | TechDawg | there are only two lines right now drbrown. |
| 01:06.23 | drbrown | not good |
| 01:06.29 | Weezey | bad dawg |
| 01:06.39 | TechDawg | I scraped the rest of the stuff as it wasn't working. |
| 01:06.41 | drbrown | that's your first prob |
| 01:06.57 | TechDawg | Incoming works fine, I'm now trying to setup outgoing. |
| 01:07.07 | drbrown | I am surprised |
| 01:07.14 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 01:07.14 | Weezey | you need more than one context |
| 01:07.46 | drbrown | outgoing, incoming, longdistance, local, zap, sip |
| 01:08.01 | drbrown | break it up as much as possible |
| 01:08.10 | TechDawg | Don't need longdistance, zap or sip |
| 01:08.12 | drbrown | makes it easier in the long run |
| 01:08.21 | drbrown | just examples |
| 01:09.04 | TechDawg | On each remote box I want two dialplans, incoming, outgoing. |
| 01:09.34 | drbrown | two contexts? |
| 01:09.58 | Bentley | O.T. - does anyone know if lame supports converting _from_ the wav49 files * writes? |
| 01:10.09 | TechDawg | The box only needs to route calls to the main * and then place calls from the main * |
| 01:11.35 | Weezey | OT: I've converted them, not automatically though. |
| 01:12.10 | TechDawg | The problem that I'm having is that unless I have a default context, the other contexts are ignored. |
| 01:12.21 | chipach | Anyone passed calls to * (with a TDM400P) from an avaya switch? The calls go through, but there is no callerid info. |
| 01:12.24 | Bentley | OT: Weezey: I'm having a brutal time finding the right lame options |
| 01:12.41 | Weezey | lemme look at my config... |
| 01:12.44 | iq | Hi, anyone faced this problem. Not all extensions can call each other. LIke 100 can call 101 and 102 and 103, etc... but 102 can not call 100... I checked the extensions.conf and everything but maybe I"m missing something. Any help? |
| 01:12.48 | Bentley | cheers |
| 01:13.23 | drbrown | iq: do you have includes in the contexts? |
| 01:13.34 | iq | drbrown, yes I do |
| 01:14.16 | Weezey | OT: -h ? |
| 01:14.46 | iq | drbrown, I do have includes in extensions.conf . anything you'd suggest to change? |
| 01:14.50 | Weezey | iq: is 100 in the same context as 102? |
| 01:14.52 | *** join/#asterisk hardwire (~hardwire@209.112.194.45) |
| 01:14.57 | bjohnson | TechDawg: your zapata.conf, sip.conf, or iax.conf are sending all incoming calls to default then |
| 01:15.05 | hardwire | I think I need to load most voice prompts into a ramdisk :) |
| 01:15.08 | iq | Weezey, yes it is all in same context, one after other |
| 01:15.08 | bjohnson | change it or just use default |
| 01:15.26 | jeffpc | what's the name of digium's FXO card? |
| 01:15.30 | TechDawg | suggestion on how to change bjohnson? |
| 01:15.31 | Weezey | iq: in sip.conf they all have the same context? |
| 01:15.34 | jeffpc | the current one |
| 01:15.39 | kb1_kanobe | using x-lite against cvs-head: when I hang up the softphone the channel stays up. What am I missing? |
| 01:15.42 | iq | Weezey, yes same context |
| 01:15.46 | hardwire | g729 is wanting to take up 20kbps :( |
| 01:16.03 | bjohnson | iq: has nothing to do with includes. make sure an exten for each phone is available in the context that each phone defaults to |
| 01:16.04 | Weezey | hardwire 8k + overhead |
| 01:16.08 | bjohnson | TechDawg: read |
| 01:16.10 | bjohnson | ~docs |
| 01:16.11 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:16.12 | hardwire | 16 ~= 20 :) |
| 01:16.16 | bjohnson | follow the examples |
| 01:16.18 | bjohnson | learn |
| 01:16.21 | hardwire | Weezey: nm |
| 01:16.22 | hardwire | haha |
| 01:16.22 | bjohnson | understand |
| 01:16.26 | TechDawg | Okay bjohnson, gotcha |
| 01:16.29 | hardwire | I thought it was 8k + 8k overhead |
| 01:16.36 | hardwire | thats still a hell of a lot of overhead.. |
| 01:16.38 | TechDawg | better to be enlighted than to be ingorant. |
| 01:16.42 | hardwire | I hope iax2 trunking helps minimize that |
| 01:16.52 | iq | bjohnson, yes boss :) |
| 01:16.59 | Weezey | hardware: less than 64k + 64k though. |
| 01:17.03 | bjohnson | jeffik: chech the digium site |
| 01:17.08 | TechDawg | That at least gets me going in the right direction, thanks bjohnson |
| 01:17.19 | iq | Weezey, thanks for help. I'm sure I would've done something wrong :) |
| 01:17.30 | drbrown | techdawg: your dial plan should have several s extension priorities |
| 01:17.42 | bjohnson | TechDawg: you can leave everything pointing to the default context .. it's no problem |
| 01:18.12 | TechDawg | That is the problem though. |
| 01:18.34 | TechDawg | Is there a way to distinguish where the incoming call is originating in the dialplan? |
| 01:18.48 | TechDawg | i.e. FXS or IAX2? |
| 01:18.55 | TechDawg | FXO or IAX2? |
| 01:19.02 | Weezey | yeah, you send incoming calls to one context and then outgoing to another. |
| 01:19.34 | Weezey | TechDawg: how many times do I have to give you the exact same advice before you go away and try something? |
| 01:20.00 | TechDawg | LOL, an IAX2 is an incoming but so is an FXO Weezey |
| 01:20.26 | TechDawg | That's where I think my loop is being caused. |
| 01:20.44 | Weezey | but you can send IAX2 from X IP to one context and FXO in (via Zap or whatever) to another. |
| 01:20.57 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 01:21.33 | TechDawg | OIC, that's where I need to go then. Unfortunately Weezey, I'll be back later when I get this one figured out. |
| 01:21.36 | TechDawg | TTFN folks |
| 01:21.39 | *** part/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 01:21.46 | Weezey | guh |
| 01:22.44 | jeffpc | did digium make a X100P replacement? |
| 01:23.12 | tzanger | jeffpc: yes |
| 01:23.24 | jeffpc | tzanger: oh? what's the name? |
| 01:23.37 | JunK-Y | TDM400P |
| 01:23.52 | jeffpc | but you need a module for that, right? |
| 01:23.58 | JunK-Y | yes |
| 01:24.09 | jeffpc | ok |
| 01:24.12 | jeffpc | thanks |
| 01:24.19 | JunK-Y | if ya wanna do the X100P job, u need FXO mod |
| 01:24.44 | tzanger | jeffpc: TDM400P with FXO modules |
| 01:25.53 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 01:26.17 | *** join/#asterisk torisa (hak@heceta.db.net) |
| 01:30.17 | *** join/#asterisk iq (~iq@65-103-167-189.omah.qwest.net) |
| 01:31.48 | kb1_kanobe | or more specifically, why does every call behave correctly except the first, which stays up regardless...? |
| 01:35.15 | drbrown | Weezey: how difficult is it to setup autoanswer with the spa-841? |
| 01:37.16 | Weezey | ez |
| 01:37.34 | *** join/#asterisk sedwards50 (~chatzilla@adsl-66-120-117-239.dsl.sndg02.pacbell.net) |
| 01:37.38 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 01:37.48 | JunK-Y | kb1_kanobe: no debug message? |
| 01:38.01 | sedwards50 | Does anybody have experience with NFAS? |
| 01:38.41 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 01:39.23 | Weezey | drbrown: see this: http://www.voip-info.org/wiki-SPA-841 specifically the auto answer section. |
| 01:39.46 | Weezey | don't forget to replace " at " with @ in the uuencoded patch to make it work. |
| 01:40.03 | iq | Weezey, bjohnson: Okay... the problem was that this user I was trying to dial was logged in for way too long. I think her registration was expired... I had her restart her ATA and it started working... Good theory or BS :) ? |
| 01:40.28 | Sato1 | cisco bought sipura? |
| 01:40.33 | JunK-Y | Sato1: read wikis. |
| 01:40.33 | Weezey | should auto-reregister |
| 01:40.36 | drbrown | I have no idea howto apply the uuencoded patch |
| 01:40.49 | Weezey | uudecode it |
| 01:41.01 | iq | Weezey, something that I can enable on client, right? |
| 01:41.03 | Weezey | then apply it like any patch -p1 < |
| 01:41.18 | drbrown | never done either |
| 01:41.19 | Weezey | you have to do that to the source code first, then re-compile |
| 01:41.30 | drbrown | I installed viat apt |
| 01:41.30 | Sato1 | JunK-Y: just got the notice, and the first thing that came to my mind was... sipura is not going to be cheap :S |
| 01:41.34 | Weezey | then * will support CALL_INFO header |
| 01:41.36 | drbrown | via |
| 01:41.40 | gambolputty | anyone get MWI working with realtime? |
| 01:42.10 | Weezey | drbrown: well, it's time to compile if you want this feature. |
| 01:42.26 | drbrown | Weezey: as I feared |
| 01:42.31 | Weezey | it's like Voice Call on Norstar system.s |
| 01:42.58 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:43.18 | drbrown | I think I will just sell snom phones for now I think |
| 01:43.21 | Weezey | works well, just don't try to include your own extension in a page. |
| 01:43.31 | iq | drbrown, what is auto answer? |
| 01:43.34 | Weezey | heh, whatever, never fear the source code. |
| 01:43.44 | drbrown | It allows for paging |
| 01:43.48 | Weezey | iq: you punch the ext and the phone just picks up. |
| 01:44.11 | iq | interesting |
| 01:44.22 | Weezey | every phone system has it. |
| 01:44.34 | drbrown | are there any phones that have as full a feature set as the snom's? |
| 01:45.48 | gambolputty | I don't think so |
| 01:45.49 | drbrown | sipura is being aquired by cisco |
| 01:45.56 | gambolputty | I have a snom 190 myself :) |
| 01:46.16 | drbrown | Like the snoms feature wise, don't like looks/price |
| 01:46.27 | Weezey | drbrown: you have a link? |
| 01:46.29 | iq | Both SPA-3000 and SPA-841 have expiration set to 3600. Is this in seconds? ...so when client sends REGISTER it can request the expiration time? |
| 01:46.41 | gambolputty | set that to 60 |
| 01:46.52 | drbrown | link??????? |
| 01:47.04 | iq | gambolputty, talking to me :) ? |
| 01:47.11 | gambolputty | yes |
| 01:47.23 | drbrown | Weezey: link??????? |
| 01:47.27 | iq | gambolputty, its in seconds, right? |
| 01:47.28 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:47.34 | gambolputty | yes |
| 01:47.43 | iq | gambolputty, thanks - will do that now :) |
| 01:47.48 | gambolputty | ok |
| 01:47.50 | Weezey | drbrown; http://www.extremetech.com/article2/0,1558,1790029,00.asp |
| 01:48.23 | iq | Linksys is too expensive... :( |
| 01:48.50 | drbrown | sorry, slooooooooooooowwwwwwwwwww tonight I guess |
| 01:48.58 | Sato1 | iq: and now sipura will be |
| 01:49.15 | Weezey | should buy some more phones before they go up |
| 01:49.21 | drbrown | Weezey: I have counted sipura out after reading that |
| 01:49.27 | iq | Sato1, :'( ...I like Sipura... got 3000 and 841... but I dont think I'll be able to afford it once it become Linksys |
| 01:49.51 | Weezey | I have 3 of each right now. |
| 01:49.59 | iq | Weezey, time to stock :) |
| 01:50.04 | newmedian | Has anyone tried the http://www.grandstream.com/y-gxp2000.htm yet? Any comments? |
| 01:50.19 | drbrown | I am looking to resell * solutions to my customers, and need feature rich phones |
| 01:50.31 | Sato1 | actually, cisco is not buying sipura, they are only doing an agreement to add voip to linksys |
| 01:50.33 | drbrown | heard grandstream had too high a return rate |
| 01:50.43 | Weezey | Sato: that's good. |
| 01:50.46 | Atacomm | Sato1: wrong, new agreement to buy them out |
| 01:51.00 | iq | Sato1, I hope you are right :) |
| 01:51.01 | Sato1 | damn! |
| 01:51.03 | MikeJ[Laptop] | drbrown, the 2000 is supposed to be better, I do not have one yet tho |
| 01:51.12 | newmedian | drbrown: LCD defects? or other issues? |
| 01:51.27 | MikeJ[Laptop] | Sato1, I think you may not be right on that |
| 01:51.42 | drbrown | newmedian: overall failure I guess, that's what my distributor said |
| 01:51.50 | iq | I'm sure this $80 2 line phones will be selling for about 150 with minor changes... changes that atleast I dont really care about |
| 01:51.51 | Atacomm | Sato1: hence the "Cisco to acquire Sipura" headline on sipura's website |
| 01:51.54 | Sato1 | i m still getting notes an news, i just got the notice a few minutes ago |
| 01:52.05 | sedwards50 | Does anybody have experience with NFAS? |
| 01:52.24 | MikeJ[Laptop] | It will spend about $68 million in cash and options for Sipura Technology |
| 01:52.52 | Sato1 | Atacomm, not headline on sipura website in my browser, thats what i m still looking for, but i m getting more information here, hehehe |
| 01:53.01 | MikeJ[Laptop] | sedwards50, not in asterisk, but yes |
| 01:53.03 | iq | Question for 841 users: Line-1 is connected to server-1 and Line-2 is connected to server-2. Can I bridge them? Like conference, etc. ? |
| 01:53.20 | bjohnson | Oh I'm a goofy goober yeah |
| 01:53.24 | drbrown | not seeing how cisco's is doing |
| 01:53.32 | drbrown | the word belly up comes to mind |
| 01:53.36 | Sato1 | sipura sales would be a little bit busy on this days |
| 01:53.38 | bjohnson | You're a goofy goober yeah |
| 01:53.39 | drbrown | stock that is |
| 01:53.42 | Atacomm | Funny thing is this is the 2nd time Cisco has bought out an ATA mfg owned and operated by the same person....its almost like it was planned; that they said okay, wanna do something cheaper, go do it, and we'll finance you and retain options to purchase you when we like...... |
| 01:54.01 | bjohnson | Were all goofy goobers yeah |
| 01:54.09 | bjohnson | Goofy goofy goober goober yeah |
| 01:54.18 | drbrown | k |
| 01:54.21 | bjohnson | nuff said |
| 01:54.26 | iq | Weezey, any idea how to do that? |
| 01:54.33 | drbrown | brb |
| 01:54.48 | bjohnson | iq: look for conference calling abilities on the 841 |
| 01:54.56 | bjohnson | likely yes |
| 01:55.01 | Sato1 | found the note at sipura site |
| 01:55.22 | iq | bjohnson, ok :) |
| 01:55.22 | bjohnson | does it say F You |
| 01:55.23 | sedwards50 | MikeJ[Laptop] -- has to asterisk, thankx. |
| 01:55.38 | iq | What is the next best thing after Sipura? Speaking of price and quality? |
| 01:55.44 | bjohnson | note on Sipura site .. F You .. we're cashing in |
| 01:55.58 | Weezey | bjohnson: too much spongebob for you mister. |
| 01:56.04 | bjohnson | kids |
| 01:56.20 | bjohnson | I even do a pretty good impression of the intro |
| 01:56.30 | bjohnson | saved me from more than one tantrum |
| 01:56.34 | Weezey | this lady at work was saying that they are all in the house singing that song |
| 01:58.44 | Weezey | bjohnson: that's a good way to sum it up. |
| 01:59.07 | newmedian | use it for your music-on-hold |
| 01:59.21 | Atacomm | why?......lol |
| 01:59.58 | mtgh | I wonder want Mark would sell it for |
| 02:00.39 | Atacomm | might want to talk to adtran first, lol |
| 02:00.47 | iq | I think Digium might go like Sipura |
| 02:00.59 | bjohnson | who cares |
| 02:01.08 | Faithful | should I / must I have qualify= in iax peer? |
| 02:01.13 | newmedian | new IAXy. when does it hit the streets? diff between v1 and v2? |
| 02:01.17 | bjohnson | it's all a game these days |
| 02:01.42 | newmedian | Faithful: that sounds like Dr. Seuss. |
| 02:01.59 | bjohnson | as it becomes more consumer oriented, they'll be all sorts of competitors |
| 02:02.22 | Faithful | newmedian: Dr. Seuss got * too? |
| 02:02.25 | bjohnson | ATAs will come down to $20 a peice and you'll be picking up SIP phones at the discount stores for $30 |
| 02:02.43 | bjohnson | I give it about 3 years |
| 02:02.52 | Faithful | bjohnson: if we ever get over our jitter |
| 02:03.25 | bjohnson | voip is getting a lot of media attention and lots of advertising $$ .. it's going mainstream and that always brings the price down significantly in the computer world |
| 02:03.33 | Faithful | newmedian: or did that mean you don't know the answer... so you made such a comment? |
| 02:03.39 | newmedian | Should you have it in a peer, should you SIP it in the clear, should you would you Qualify |
| 02:03.51 | key2 | where do I configure to what context to go when I place a call from an SIP phone ? |
| 02:04.00 | bjohnson | Faithful: I like qualify=yes .. but it interferes with some providers |
| 02:04.13 | newmedian | Was being capricious rather than helpful, sorry. |
| 02:04.24 | *** join/#asterisk Shido6 (~shido@d57-87-253.home.cgocable.net) |
| 02:04.28 | Faithful | newmedian: ;-) |
| 02:04.29 | bjohnson | newmedian: not to do it, I don't know why |
| 02:04.30 | key2 | in global or in [MYSIP] |
| 02:04.40 | JunK-Y | MYSIP |
| 02:04.47 | JunK-Y | for that specific sip account |
| 02:05.32 | Faithful | Well I am having a rather strange thing happen which I have finally figured out... after some time 1day - 2days or maybe just hour my peers become unreachable with iax poke |
| 02:05.39 | key2 | JunK-Y: and then no matter what number I call frop this SIP it would go to that context ? |
| 02:05.40 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
| 02:05.51 | JunK-Y | yes |
| 02:06.25 | key2 | JunK-Y: and in this context, how would I specify that no matter what number has been dialed, do that? with _.,1,dothat() ??? |
| 02:06.36 | iq | anyone got a hack for spa-841 to make it 4 lines :) ? |
| 02:07.12 | Weezey | iq: you pay $30 and get the other two lines |
| 02:07.17 | Weezey | it's a great hack |
| 02:07.26 | Sato1 | hehehehe |
| 02:07.40 | key2 | JunK-Y: and in this context, how would I specify that no matter what number has been dialed, do that? with _.,1,dothat() ??? |
| 02:08.13 | Weezey | key2: you press up arrow, enter again. |
| 02:08.16 | iq | Weezey, I'm broke |
| 02:08.45 | Weezey | iq: fundraising. |
| 02:09.29 | key2 | sorry :) |
| 02:09.51 | newmedian | you were showing him the arrow of his ways? |
| 02:09.53 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 02:10.02 | Weezey | man, you're lame tonight. |
| 02:10.06 | Weezey | :) |
| 02:10.43 | jpayne | anyone having problems making outbound calls to nufone ? |
| 02:12.38 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 02:13.33 | key2 | Weezey: so how do I specify what should be done when i call from a specific SIP ? like if I call any number from SIP 201, I want him to playback a sound, how should I do that |
| 02:13.51 | *** join/#asterisk apardo (~apardo@80-28-161-183.adsl.nuria.telefonica-data.net) |
| 02:13.56 | apardo | hi. Searching for IAX2 peers in wireless community networks: http://velouria.sysnetworks.net/archives/2005/05/05/111/ |
| 02:15.17 | rvhi | did anyone try the polycom latest version 1.5.2? |
| 02:18.39 | puowvip | time to sleep |
| 02:18.51 | JunK-Y | rvhi: nah |
| 02:21.02 | key2 | JunK-Y: so how do I specify what should be done when i call from a specific SIP ? like if I call any number from SIP 201, I want him to playback a sound, how should I do that |
| 02:22.28 | newmedian | key2, I think you need to do a bit more reading. :) But are you thinking along these lines? http://pastebin.ca/11003 |
| 02:23.09 | Inv_arp | bah BV just stopped working... ok thats it iax.cc time |
| 02:24.11 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 02:24.12 | MikeJ[Laptop] | key2, newmedian's example does what you are saying, just make sure the context= for that peer is set to the context your dialplan for calls from that user go... |
| 02:24.33 | Inv_arp | ok hmm how to cx BV |
| 02:25.54 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 02:28.19 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:32.08 | Faithful | Well I am having a rather strange thing happen which I have finally figured out... after some time 1day - 2days or maybe just hour my peers become unreachable with iax poke ... but if I delete my arp entries on the * box I can call again... anyone??? |
| 02:32.15 | bjohnson | puowvip: no time for that |
| 02:32.31 | Faithful | well in particular my arp entry to the gateway |
| 02:32.53 | bjohnson | key2: all standrd stuff with hundreds of examples |
| 02:33.09 | *** join/#asterisk SuperN (SuperN@168satb3.codetel.net.do) |
| 02:33.15 | SuperN | good night everyone! |
| 02:33.20 | bjohnson | Faithful: sounds like a general network problem then |
| 02:33.33 | Faithful | it ain't general |
| 02:33.43 | bjohnson | ie .. not specific to asterisk |
| 02:33.51 | Faithful | but it is a problem |
| 02:34.04 | bjohnson | can you ping or otherwise connect to the gateway when asterisk can't? |
| 02:34.09 | SuperN | I need help to setup a h323 channel |
| 02:34.24 | bjohnson | SuperN: wow .. not from me |
| 02:34.26 | Faithful | bjohnson: * can still ping the hosts |
| 02:34.37 | bjohnson | the gateway you said was the problem |
| 02:35.10 | Faithful | bjohnson: no I said deleting the arp entry remedies it for a time |
| 02:35.13 | SuperN | have you ever use a Netphone ke1020a with asterisk? |
| 02:35.47 | bjohnson | Faithful: I'm afraid I have no idea .. try the mailing list with an enticing subject line |
| 02:35.52 | *** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net) |
| 02:36.05 | SuperN | ok :( |
| 02:37.53 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 02:38.23 | rvhi | what's the maximum number of context that * can support? |
| 02:38.31 | shido6 | hehehe |
| 02:38.36 | shido6 | isnt that the question :) |
| 02:38.39 | SuperN | Can I use the asteriks with a Winmodem? |
| 02:38.46 | shido6 | its really a matter of dialplan logic I find. |
| 02:38.50 | Inv_arp | quick which does anyone prefer Voicepulse or iax.cc? |
| 02:39.17 | JunK-Y | rvhi: more then ya can use :) |
| 02:40.52 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-167.modem.logical.net) |
| 02:40.59 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 02:43.49 | bjohnson | rvhi: no one has found a limit |
| 02:45.43 | key2 | once I changed the extansion.conf, how do I tell asterisk to reload it / |
| 02:45.43 | key2 | ? |
| 02:45.56 | mtgh | at the cli |
| 02:45.58 | mtgh | run reload |
| 02:47.48 | *** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 02:48.33 | bjohnson | "reload" |
| 02:48.38 | bjohnson | forget the run |
| 02:49.31 | MikeJ[Laptop] | ~docs |
| 02:49.44 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 02:49.44 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 02:50.04 | bjohnson | doh .. I wondered where I left that |
| 02:51.25 | MikeJ[Laptop] | your docs... |
| 02:51.30 | MikeJ[Laptop] | ? |
| 02:56.06 | SuperN | can I use asterisk as a soft switch? |
| 02:56.17 | file | sure, why not |
| 02:56.22 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:57.53 | SuperN | file: where can I find info about it? |
| 02:58.01 | file | SuperN: well what do you want to do exactly? |
| 02:58.40 | bjohnson | rule the world |
| 02:58.44 | SuperN | the problem is: the netphone I have here says: that operates in a soft switch platform |
| 02:58.59 | file | bjohnson: that's in app_ruleworld |
| 02:59.07 | file | bjohnson: it takes the planet name as the argument |
| 02:59.19 | file | SuperN: well what protocol does it use? you need to know in order to use it... |
| 02:59.19 | bjohnson | SuperN: that's marketing |
| 02:59.29 | bjohnson | he was asking about h323 earlier |
| 02:59.35 | file | oh dear god |
| 02:59.49 | SuperN | yeah, becuase I trying to figure out.. how to put this phone to work |
| 02:59.52 | bjohnson | I remember him in my prayers |
| 03:00.51 | SuperN | this phone, does not have any documentation.. just what I am asking in this channel |
| 03:00.52 | file | no, that would cause blindness |
| 03:01.13 | bjohnson | bad purchase decision |
| 03:01.20 | SuperN | :( |
| 03:01.26 | bjohnson | you're going to spend hours on it |
| 03:01.33 | bjohnson | maybe longer |
| 03:01.44 | SuperN | I din't buy it .. my client did |
| 03:01.48 | SuperN | :( |
| 03:01.56 | file | and in the end become frustrated, say we're useless, and storm out of the channel in a rage |
| 03:02.06 | bjohnson | what are you charging him per hour and how many phones? |
| 03:02.54 | SuperN | nothing by now.. becuase I am newbie using asterisk.. I need to offer him a low cost solution ... |
| 03:03.06 | bjohnson | tada .. upgrade to SIP |
| 03:03.08 | SuperN | using what he has now |
| 03:03.12 | bjohnson | firmware upgrade |
| 03:03.33 | SuperN | you mean for the phone? |
| 03:03.47 | bjohnson | yes .. firmware and everything else |
| 03:03.59 | SuperN | I did it! |
| 03:04.06 | bjohnson | you can get a new phone for like $80 that is a true SIP phone |
| 03:04.15 | *** join/#asterisk ShadowMaster1 (~askme@host89-133.rancor.birch.net) |
| 03:04.40 | SuperN | can I use sip software to test asterisk I think... ? |
| 03:04.49 | file | SuperN: sure. |
| 03:04.55 | SuperN | ok |
| 03:05.09 | SuperN | because I have a laptop here.. so I can test the asteriks configuration |
| 03:05.11 | ShadowMaster1 | I've used the X-Lite softphone on Windows with good success |
| 03:05.15 | *** join/#asterisk pbx123 (~nbrand@adsl-69-230-199-226.dsl.irvnca.pacbell.net) |
| 03:05.20 | file | eek a master of the shadow |
| 03:05.21 | SuperN | I was trying the Dial stuff and I think it was all right |
| 03:06.15 | SuperN | I have the oh323.conf file in the /etc/asterisk |
| 03:06.26 | SuperN | but the asterisk does not load that config file |
| 03:06.40 | SuperN | I have a Suse 9.3 O.S. installed.. |
| 03:06.44 | xeet2 | is anyone here a dundi peer? |
| 03:06.50 | file | xeet2: yesssssss |
| 03:07.19 | ShadowMaster1 | Can I get some hlp with what might be more a linux question then an Astrisk question? I had everything working great, till I rebooted. How do I get Astrisk to reload when ever the system restarts? |
| 03:07.26 | ShadowMaster1 | runing Debian.. |
| 03:07.38 | SuperN | asterisk -vvvvvc |
| 03:07.49 | ShadowMaster1 | but that quits as soon as I close the session. |
| 03:08.30 | ShadowMaster1 | maybe not... Ok, where do I need to put that so it loads automaticly at boot up? |
| 03:08.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 03:08.36 | pjz | ShadowMaster1: first of all, look in /usr/share/doc/asterisk |
| 03:08.37 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.175) |
| 03:08.42 | SuperN | if you type just "asterisk" it will run as a service |
| 03:08.43 | Carp1 | Whats a good FTP server I can install with Yum? |
| 03:08.48 | pjz | ShadowMaster1: and zmore the README.Debian.gz you find there |
| 03:08.51 | SuperN | you should use the rc directories |
| 03:08.54 | pjz | ShadowMaster1: and it will tell you.. |
| 03:09.11 | sudhir492 | mpg123 makes my brandnew asterisk crash :-) |
| 03:09.11 | SuperN | I don't remember how to do it.. but I know it's there |
| 03:09.12 | pjz | ShadowMaster1: ..that you need to do the obvious edit in /etc/defaults/asterisk |
| 03:09.15 | sudhir492 | :-( |
| 03:09.23 | ShadowMaster1 | I type Asterisk, and notrhing happens.. I then try the asterisk -r and it says Unable to connect to remote asterisk |
| 03:09.43 | sudhir492 | Carp1: try vsftp |
| 03:09.52 | sudhir492 | stands for very secure ftp |
| 03:11.37 | pjz | ShadowMaster1: in /etc/default/asterisk, set RUNASTERISK="yes" |
| 03:11.52 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:12.10 | ShadowMaster1 | yes, I just found that one.. Not sure how it got changed to no, but it's yes now... thanks.. |
| 03:12.18 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 03:12.21 | JunK-Y | ShadowMaster1: asterisk -c ? |
| 03:12.22 | pjz | ShadowMaster1: that gets included by /etc/init.d/asterisk |
| 03:12.30 | ShadowMaster1 | Next question: How do I get the zaptel drivers to autoload (or will they when Asterisk starts?) |
| 03:12.33 | pjz | ShadowMaster1: so now you can /etc/init.d/asterisk start |
| 03:16.49 | ShadowMaster1 | before Asterisk is loaded, and I do a ps -all, should I see the wcfxo and zaptel drivers? |
| 03:17.04 | Qwell | lsmod |
| 03:17.18 | Damin | ShadowMaster1: Use /usr/src/zaptel/zaptel.init. Put it into your /etc/rc.d/init.d/ directory |
| 03:17.26 | Qwell | and yes, probably, unless maybe you're running udev...I don't know in that case |
| 03:17.38 | tzafrir_laptop | lsmod |grep zaptel |
| 03:18.07 | Damin | ShadowMaster1: And no, ps will not show the zaptel drivers. They are kernel modules.. |
| 03:18.19 | ShadowMaster1 | lsmod does show it (and it's dependence on wcfxo).. |
| 03:18.32 | Damin | ShadowMaster1: That means zaptel is loaded.. |
| 03:18.35 | ShadowMaster1 | damin: Ok, that makes sense.. zaptel being kernel.. no prob.. |
| 03:19.04 | tzafrir_laptop | ShadowMaster1, wcfxo depends on zaptel, not the other way around |
| 03:19.34 | ShadowMaster1 | Yes, I'm sorry, I typed it backwards... |
| 03:19.58 | ShadowMaster1 | and the lsmod shows the wcfxo module as well. |
| 03:20.22 | tzafrir_laptop | ShadowMaster1, if you use Debian, do use the init.d script from the distro and not the default zaptel.init |
| 03:20.55 | tzafrir_laptop | That zaptel.init probably uses some functions not avilable in debian anyway. |
| 03:21.41 | tzafrir_laptop | On a standard debian sarge/sid installation, loading the modules usually happes by discover. |
| 03:22.02 | tzafrir_laptop | If that doesn't work (or you don't like it) add the module name to /etc/modules |
| 03:24.02 | ShadowMaster1 | it still doesn't look like zaptel is working.. when I do the asterisk start I get warnings and errors:: See http://pastebin.ca/11007 |
| 03:24.20 | Damin | tzafrir: The zaptel.init that is in head (which I just patched and submitted Bug #4176) understands Debian and works fine with it. |
| 03:25.53 | tzafrir_laptop | Damin, what I meant is that loading the modules is already sone elsewhere. All that needs doing is running ztcfg, if it has not been done at modprobe time |
| 03:26.07 | Damin | tzafrir: Yes. |
| 03:26.09 | ShadowMaster1 | I did a locate and found 2 on my system at /usr/downloads/zaptel-1.0.7/zaptel.init, and /usr/src/zaptel/zaptel.init |
| 03:27.34 | tzafrir_laptop | Damin, there is another issue: if you have several types of modules, discover (plug&play loader) may load them in a different order than the order you've loaded them manually |
| 03:27.46 | *** join/#asterisk simon556 (~sam@211.25.2.30) |
| 03:27.52 | simon556 | hi guys |
| 03:27.58 | simon556 | i m simon from malaysia |
| 03:28.09 | *** part/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 03:28.16 | simon556 | anyone can help me about IAX2 trunking |
| 03:28.20 | torisa | hi simon from malaysia. we're #asterisk |
| 03:28.32 | ShadowMaster1 | tzafrir: I don't see them in the modules config. What do I need to add there? (Sorry, if I seem dense -- I'm learning Linux as I go along here) |
| 03:28.38 | simon556 | we are asterisk |
| 03:28.51 | simon556 | i m using asterisk 2 |
| 03:29.02 | ShadowMaster1 | and I'm trying to .. <grin> |
| 03:29.04 | simon556 | how to set iax2 trunking in asterisk |
| 03:29.18 | simon556 | connecting 2 servers |
| 03:31.12 | Qwell | asterisk 2? |
| 03:31.17 | opus_ | 2.0 baby |
| 03:31.29 | Qwell | wow, this cat is from the future! |
| 03:31.35 | opus_ | yeah |
| 03:31.44 | Qwell | "we are asterisk" |
| 03:31.48 | Qwell | and he's a borg |
| 03:31.55 | opus_ | yup. |
| 03:31.55 | Qwell | s/a // |
| 03:32.47 | opus_ | simon you must assimulate both servers |
| 03:32.58 | drbrown | has anyone setup an ivr before? |
| 03:33.02 | Qwell | showed him... |
| 03:33.05 | opus_ | damn |
| 03:33.08 | Qwell | NEXT |
| 03:33.19 | Qwell | drbrown: Whats the problem? |
| 03:33.38 | drbrown | File greeting does notexist in any format |
| 03:33.46 | drbrown | that's the error it gives |
| 03:33.51 | drbrown | no playback |
| 03:33.52 | Qwell | Do you have greeting? |
| 03:33.56 | drbrown | yes |
| 03:34.05 | Qwell | Where? |
| 03:34.06 | *** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca) |
| 03:34.07 | drbrown | do I have to designate the file extension? |
| 03:34.10 | Qwell | no |
| 03:34.19 | Qwell | Where is your file located? |
| 03:34.31 | drbrown | "/var/lib/asterisk/sounds" |
| 03:34.34 | opus_ | cp `find / -name '*.gsm'` /usr/lib/asterisk/sounds/ |
| 03:34.38 | Qwell | What are the perms on it? |
| 03:34.44 | Qwell | Can the user running asterisk read it? |
| 03:35.09 | Qwell | opus_: -type f |
| 03:35.10 | Qwell | safer... |
| 03:35.12 | drbrown | I have changed the ownership to the group asterisk, and the user can read it |
| 03:35.16 | asterisk99 | How can I install Zaptel without using asttinstall (I use Gentoo and the installer says it's not compatible)??? |
| 03:35.28 | drbrown | Maybe the sounds are in the wrong spot |
| 03:35.38 | opus_ | what file format is it |
| 03:35.44 | drbrown | gsm |
| 03:35.55 | opus_ | is it 8k? |
| 03:35.58 | drbrown | I am using the how to on the wiki |
| 03:36.03 | opus_ | won't work at 22k |
| 03:36.08 | drbrown | not sure |
| 03:36.11 | opus_ | i think, dunno. |
| 03:36.23 | drbrown | i am recording it with asterisk |
| 03:36.28 | opus_ | oh |
| 03:36.56 | opus_ | yeah, just find the file and put it in /usr/lib/asterisk/sounds/<NAME_OF_SOUND>.gsm |
| 03:37.02 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 03:37.20 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 03:37.27 | opus_ | and then play <name_of_sound> in you dial plan... |
| 03:37.57 | drbrown | only modules in "/usr/lib/asterisk" |
| 03:38.31 | opus_ | backup /etc/asterisk real quick, go to /usr/src/asterisk and "make samples", then... unbackup |
| 03:38.55 | drbrown | can't do that |
| 03:39.22 | drbrown | did not build from source |
| 03:39.25 | drbrown | apt |
| 03:39.26 | opus_ | then cp -favr /usr/src/asterisk/sounds /usr/lib/asterisk/sounds/ |
| 03:39.30 | drbrown | deb package |
| 03:39.32 | opus_ | get the source |
| 03:39.53 | opus_ | screw the package |
| 03:40.04 | ShadowMaster1 | try apt-get install asterisk-sounds |
| 03:40.05 | drbrown | I would rather use the package |
| 03:40.22 | asterisk99 | anyone know how to manually install zaptel? |
| 03:40.26 | opus_ | Actually, has anyone ever got debian MIPS processor binaries to run on a linksys wrt54g? |
| 03:40.35 | tzafrir_laptop | what's wrong with the deb package? |
| 03:40.39 | tzafrir_laptop | missing moh? |
| 03:40.56 | drbrown | asterisk-sounds-main is already the newest version. |
| 03:41.05 | tzafrir_laptop | opus_, http://packages.debian.org/package_name |
| 03:41.19 | tzafrir_laptop | as for the moh files: |
| 03:42.23 | kb1_kanobe | i'm going insane trying to get the 'multiple peers' model for codec selection going - should I be specing 'disallow=all' for each as well as 'allow=preferedcodename', or should I just have the 'allow=...' directive? |
| 03:42.27 | opus_ | asterisk99 -- cd /usr/src/zaptel && .. edit /etc/zaptel.conf make && make install && modprobe zaptel && modprobe wcfxo etcetc ... /sbin/ztcfg -vv |
| 03:42.34 | tzafrir_laptop | http://tzafrir.org.il/rapid/asterisk-sounds-moh_1.0-1_all.deb |
| 03:42.54 | tzafrir_laptop | Those moh files really don't belong in the main asterisk package |
| 03:44.21 | tzafrir_laptop | drbrown, what sounds are you missing? |
| 03:44.25 | asterisk99 | opus: astinstall didn't create /usr/src/zaptel |
| 03:44.36 | drbrown | only the ones I have recorded |
| 03:44.42 | drbrown | All the other sounds work |
| 03:44.49 | drbrown | voicemail works |
| 03:44.56 | *** join/#asterisk juice (~juice@mo-65-173-76-11.dyn.sprint-hsd.net) |
| 03:45.07 | tzafrir_laptop | you mean the ones that don't come with the asterisk tarball itself? |
| 03:45.10 | drbrown | everything with the exception of my home brew sounds |
| 03:45.18 | drbrown | made them from within asterisk |
| 03:45.38 | drbrown | when playing a sound is it nessesarry to put in the gsm file extension? |
| 03:46.04 | drbrown | tzafrir_laptop: yes |
| 03:48.42 | opus_ | use the source |
| 03:49.09 | tzafrir_laptop | drbrown, look at /usr/share/asterisk/sounds |
| 03:49.54 | *** part/#asterisk at561 (~aoiahsdf@68.71.213-37.atlsfl.adelphia.net) |
| 03:49.56 | tzafrir_laptop | drbrown, no, you don't need to put .gsm in gsm files |
| 03:50.39 | drbrown | eureka |
| 03:51.02 | drbrown | "/usr/share/asterisk/sounds" |
| 03:51.18 | drbrown | thanks guys |
| 03:51.24 | drbrown | apreciate it |
| 03:51.48 | drbrown | the only reason I don't want to use the source, is because this is a production system |
| 03:52.08 | Vco | ? |
| 03:52.56 | elric | drbrown, even more of a reason to use source. |
| 03:53.33 | drbrown | elric, not comfortable yet |
| 03:53.56 | drbrown | elric, also complie asterisk on a p1 200 |
| 03:54.06 | drbrown | takes to long for my liking |
| 03:54.16 | elric | drbrown, read the apt-get manual, i think you can use apt to compile, if i am not mistaken |
| 03:54.33 | elric | drbrown, i use low end hardware too, leave it over night |
| 03:54.55 | drbrown | erlric, interesting never thought of that |
| 03:55.28 | drbrown | elric, the deb package format is the only reason I switched to debian for my asterisk servers |
| 03:58.08 | *** join/#asterisk SuperN (SuperN@127satb3.codetel.net.do) |
| 03:58.18 | SuperN | file |
| 03:58.32 | SuperN | I am back.. I got the chipset the phone is using |
| 03:59.22 | MikeJ[Laptop] | mmmmmmm |
| 03:59.36 | SuperN | I have a PA1688 chipset phone |
| 03:59.53 | SuperN | they are supposed to support SIP RFC 2543 AND RFC3261 |
| 04:01.59 | remmo | supern: they are nice cheap phones |
| 04:02.26 | SuperN | remmo: I can't put the one I have to work with asterisk |
| 04:02.31 | SuperN | do you any config example? |
| 04:02.40 | remmo | yeah |
| 04:02.52 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 04:03.02 | remmo | first upgrade the firmware from www.aredfox.com sip or iax2 is fine. |
| 04:03.31 | remmo | 1.42 is great for iax |
| 04:03.51 | SuperN | but the phone comes with a feature to do it from it |
| 04:04.07 | tainted- | . |
| 04:04.15 | SuperN | I have ver 2.19.00 |
| 04:04.16 | remmo | not mine i have only used the 5111 version |
| 04:04.24 | remmo | whats the vendor? |
| 04:04.46 | SuperN | 03-14-2005 |
| 04:04.46 | SuperN | is the date |
| 04:04.47 | SuperN | NetPhone KE1020A |
| 04:04.52 | SuperN | that's all the phone says |
| 04:05.19 | remmo | oh, that might be only compatiable with netphone. not sure. i only know that you need the right firmware 1st |
| 04:06.07 | SuperN | do you have any config file example for your phone? |
| 04:06.22 | remmo | yeah what are you after sip or iax? |
| 04:06.36 | SuperN | sip |
| 04:07.33 | SuperN | I am updating the phone using his own feature for this, right now |
| 04:08.11 | *** join/#asterisk `Sauron (sauron@rrcs-24-153-164-117.sw.biz.rr.com) |
| 04:12.24 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 04:13.38 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 04:14.15 | *** join/#asterisk drumkilla (~russell@pcp03785394pcs.mplsnt01.sc.comcast.net) |
| 04:14.15 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 04:16.40 | *** join/#asterisk meppl (mephisto@p54AACE0A.dip.t-dialin.net) |
| 04:18.38 | lyoungz | I'm having voice-quality problems recording conversations with asterisk |
| 04:19.17 | lyoungz | when tying asterisk with a Digium card to a Nortel PBX's analog port |
| 04:19.55 | lyoungz | the voice is recorded, but is very filled with static and broken |
| 04:21.21 | dudes | When is a phone recording actually good? But perhaps they have a different setting for a high quality format. |
| 04:22.06 | lyoungz | It is bad in a way that suggests a sampling problem, analogous to a moire pattern in graphics. |
| 04:22.21 | lyoungz | it is too consistently awful to be merely poor quality |
| 04:22.26 | kb1_kanobe | lyoungz: probably a timing problem - have you got the clock source set correctly on the DTI relative to zaptel.conf? |
| 04:22.47 | kb1_kanobe | can anyone explain why, when I use 'allow=gsm' for the codec for an iax peer the 'prefered codec' that's passed is slin? |
| 04:23.24 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 04:23.25 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 04:23.50 | lyoungz | kb1_kanobe: my zaptel.conf doesn't have a clock source listed. |
| 04:23.51 | dudes | because it's allowed and preferred in the configs, perhaps? |
| 04:24.33 | kb1_kanobe | err... how should I be specifying my prefered codec then? I only want GSM on that peer. |
| 04:24.49 | lyoungz | kb1_kanobe: I just have fxo_ks=1 fxs_ks=4 loadzone=us defaultzone=us |
| 04:25.03 | kb1_kanobe | is this a t1 interface? |
| 04:25.35 | dudes | kb1_kanobe - disallow=all allow=gsm |
| 04:25.41 | kb1_kanobe | no... analog. My dumb. |
| 04:25.43 | lyoungz | kb1_kanobe: no, it is analog. The nortel supports analog handsets, so I connected a Digium analog port to the handset port |
| 04:26.02 | kb1_kanobe | mmm.. dunno then, sorry. |
| 04:26.11 | kb1_kanobe | (wasn't paying enough attention) |
| 04:26.15 | lyoungz | kb1_kanobe: ok, thanks for the try |
| 04:27.24 | dudes | lyoungz - so the nortel PBX is doing the recording or is asterisk? |
| 04:29.56 | kb1_kanobe | dudes - this is cvs-head. When I specify as you've suggested for the peer entry the dial fails, badly with a 'raw hangup debug' on the peer. |
| 04:30.34 | kb1_kanobe | here it is - on the originating machine: iax2_request: Unable to create translator path for unknown to slin |
| 04:30.57 | dudes | kb1_kanobe - perhaps the host isn't setup to use gsm |
| 04:31.20 | kb1_kanobe | the 'user' entry on the other host is 'allow=all' |
| 04:31.59 | dudes | I'd just disallow all and allow= the codecs I'd preferr |
| 04:32.08 | dudes | personally. |
| 04:32.26 | kb1_kanobe | duh, entry order. I pasted disallow after allow. |
| 04:32.28 | kb1_kanobe | I need more sleep. |
| 04:32.43 | dudes | join the club =p |
| 04:44.51 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 04:50.31 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 04:51.11 | *** part/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 04:52.27 | kb1_kanobe | is it possible to specify a 'debounce' time to the iax 'qualify' directive? Ie. if latency exceed qualify for this host then mark unreachable for n seconds minimum? |
| 04:55.17 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 04:56.34 | ManxPower | kb1_kanobe: not that I know of. |
| 04:59.07 | kb1_kanobe | how about an iax user context that is basically 'allow all'? |
| 05:01.05 | wisdom | kb1_kanobe Yes, it's possible to do, but not with asterisk directly, would have to use external scripts to do the checking and then have another script that generates your dialplan to eliminate the use of that peer until the script changes the status and recreates the config again |
| 05:01.28 | wisdom | kb1_kanobe That's not a "bad" thing, provided the only thing they can reach are your public extensions |
| 05:01.34 | wisdom | (the IAX stuff) |
| 05:03.18 | kb1_kanobe | this is an entirely private network - seperate cabling. |
| 05:07.53 | *** join/#asterisk mwgbc (mwallacegb@adsl-69-109-163-121.dsl.pltn13.pacbell.net) |
| 05:08.36 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 05:08.47 | mwgbc | I am having problems with a Asterisk Win32 load, it is not recognizing DTMF. Any ideas? |
| 05:09.51 | drumkilla | don't run windows |
| 05:10.15 | wisdom | learn to love unix |
| 05:10.46 | dudes | that's what I was going to say, but didn't want to piss anyone off =p |
| 05:10.47 | mwgbc | drumkilla : Thanks, for the obvious :-) I use Linux primarily, but I have a propriatary Voice Broadcast program that needs windows |
| 05:11.31 | *** join/#asterisk Tond (Tond@Toronto-HSE-ppp3646966.sympatico.ca) |
| 05:11.39 | wisdom | wine has it's uses :) |
| 05:11.40 | dudes | mwgbc - find an open-source variety |
| 05:11.45 | Tond | Has anyone compiled the G729 codec for firefly? |
| 05:12.12 | Tond | I don't have Visual C++ installed here and I need to compile their source code to get the g729.dll file |
| 05:12.23 | mwgbc | I'm locked into the use of this (boss says so) but I havnt tried wine |
| 05:12.46 | opus_ | tond send it to me i'll do it |
| 05:12.54 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 05:13.07 | libpcp | hi all |
| 05:13.13 | opus_ | mwgbc try linux |
| 05:13.30 | drumkilla | mwgbc, we already went over this |
| 05:13.34 | drumkilla | :p |
| 05:13.36 | Tond | Thanks Opus_ |
| 05:13.59 | libpcp | i have digium 4 ports fxo, is it possible to group those 4 channels and call in dialplan for dialout? |
| 05:14.53 | mwgbc | Well, thanks anyway. I'm not looking for the perfect solution, or I would use Linux, I'm tring to make the best of a bad situation as that's what I'm stuck with. |
| 05:15.06 | mwgbc | bye all |
| 05:15.18 | *** part/#asterisk mwgbc (mwallacegb@adsl-69-109-163-121.dsl.pltn13.pacbell.net) |
| 05:17.12 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:19.06 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:19.22 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 05:23.13 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 05:23.44 | |Vulture| | Anyone here ever configure a Layer 2 switch so that only specified MACs can use the switch? |
| 05:25.43 | *** join/#asterisk MKnight (~ATsirogia@S0106000f665a41db.vf.shawcable.net) |
| 05:26.18 | MKnight | anyone know what the hardware requirments are to run asterisk? The official site doesn't specify anything |
| 05:26.27 | |Vulture| | MKnight: depends on your setup |
| 05:26.35 | wisdom | depends on the number of active calls and what it's going to be doing |
| 05:26.36 | |Vulture| | MKnight: what is your plan? |
| 05:27.23 | MKnight | its for soho use |
| 05:27.41 | wisdom | then a P200 would do :) |
| 05:27.54 | wisdom | but if you have any decent PIII's it makes things go a tad quicker :) |
| 05:27.57 | MKnight | i only have a spare P4 2.6 here |
| 05:28.05 | MKnight | with 2GB of ram |
| 05:28.15 | wisdom | then it's a little overkill, but it'd make due :) |
| 05:28.15 | MKnight | what kind of use can I expect from that? |
| 05:28.39 | |Vulture| | MKnight: anything soho can throw at it |
| 05:28.50 | MKnight | I'm on a 5mbps down/1mbps up cable connection |
| 05:28.58 | wisdom | more than enough |
| 05:29.05 | |Vulture| | 1mbps up... lucky bastard |
| 05:29.06 | MKnight | thanks |
| 05:29.14 | MKnight | lol, its not enough still |
| 05:29.23 | wisdom | I can't wait for FIOS |
| 05:29.31 | wisdom | 30 down, 5 up |
| 05:29.39 | wisdom | it's 2 blocks away |
| 05:29.39 | MKnight | I only get 50GB of month combined download/upload use though |
| 05:29.41 | Qwell | wisdom: They dug a trench in my coworkers yard today. :( |
| 05:29.47 | MKnight | my ISP is very strict with bandwidth use |
| 05:29.57 | wisdom | Qwell Well, that's a good thing for your coworker :) |
| 05:30.07 | |Vulture| | MKnight: then that might suck for voip depending on how much you do |
| 05:30.14 | wisdom | MKnight they'll loosen those requirements up once they get some competition |
| 05:30.20 | |Vulture| | wisdom: is there a site that show its availability? |
| 05:30.36 | wisdom | |Vulture| Soho use won't exceed their thresholds, run a decent size call center and you will :) |
| 05:30.39 | MKnight | what nix distribution should i run with asterisk? |
| 05:30.46 | Qwell | |Vulture|: if Verizon has been in your neighborhood digging up the street and laying fiber...you'll know when its coming |
| 05:30.54 | MKnight | I want somethign stable |
| 05:31.07 | |Vulture| | wisdom: I run everything over PRIs and just a small ammount over voip now |
| 05:31.13 | |Vulture| | Qwell: we have Bellsuck here |
| 05:31.31 | Qwell | Then you aren't getting FIOS. :p |
| 05:31.45 | |Vulture| | :( |
| 05:32.01 | |Vulture| | hmm anyone familliar with Netgear L2 Switches? |
| 05:32.09 | MKnight | alright, so all I need is a modem and an ATA then I guess |
| 05:32.11 | Qwell | I'm gonna be so incredibly pissed if Verizon forces you to have a landling to get FIOS |
| 05:32.15 | wisdom | |Vulture| http://www.verizonfios.com |
| 05:32.18 | Qwell | landline* |
| 05:32.22 | |Vulture| | they have a "MAC Address Lockdown Table" but I cant figure out what it does |
| 05:32.50 | opus_ | i can't wait until my neighbor wireless network gets it as well |
| 05:32.51 | opus_ | !! :) |
| 05:33.03 | |Vulture| | opus_: lol |
| 05:33.31 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 05:35.03 | |Vulture| | anyone ever have a Polycom IP500 that just randomly looses connection? Doesn't say "Network cable disconnected" Ive tried multiple ports/cables |
| 05:35.21 | |Vulture| | just loses the connection and the output connection to the PC |
| 05:35.31 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:37.52 | *** join/#asterisk cc (~cc@byte.fedora) |
| 05:39.26 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 05:42.23 | opus_ | vulture - umm, which firmware are you using |
| 05:42.30 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:43.12 | |Vulture| | not the newest the one before that... but I use it on like 50 phones |
| 05:43.18 | |Vulture| | 2.5.0 with 3.4 or something |
| 05:47.17 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:49.20 | *** join/#asterisk gaba (~gaba@protest.net) |
| 05:52.35 | *** join/#asterisk iswm (iswm@iswm.user) |
| 06:05.00 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 06:10.48 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:13.07 | opus_ | is broadvoice currently down? |
| 06:17.02 | opus_ | I think it is down... |
| 06:17.31 | opus_ | damn, they have the worst fucking service ever. |
| 06:17.38 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 06:17.42 | |Vulture| | opus_: been down for awhile actually |
| 06:17.47 | |Vulture| | for inbound anyways |
| 06:17.55 | opus_ | really |
| 06:18.04 | opus_ | so, like, what are they doing with my money:) |
| 06:18.33 | opus_ | vulture - do you have a 50 phone IP500 asterisk install? |
| 06:19.04 | Qwell | opus_: They're putting it to good use, I assure you |
| 06:19.13 | Qwell | That pub is getting alot of business as of late |
| 06:19.18 | opus_ | haha |
| 06:19.19 | Qwell | :) |
| 06:19.42 | opus_ | i administered a 50+ ip500 install and had to RMA about 3 phones a month |
| 06:20.23 | opus_ | most of all went on ebay for $80-120 bucks |
| 06:26.04 | bsdfreak | what's nuphone's urk? |
| 06:26.06 | bsdfreak | er.. url |
| 06:26.43 | bsdfreak | are they dead? |
| 06:27.52 | ManxPower | www.nufone.net |
| 06:28.07 | bsdfreak | hmm |
| 06:28.10 | bsdfreak | doesn't seem to resolve. :x |
| 06:28.57 | ManxPower | they are doing upgrades |
| 06:29.01 | bsdfreak | oh. |
| 06:29.01 | opus_ | the internet is down |
| 06:29.11 | ManxPower | the web site will be the last to get working, I'm sure. |
| 06:29.17 | bsdfreak | ah well. |
| 06:29.21 | bsdfreak | i'll sign up later. |
| 06:30.19 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 06:30.33 | lehel | hello |
| 06:30.41 | opus_ | i guess i know when broadvoice is down is when there own number is down.. haha |
| 06:30.49 | *** join/#asterisk Jas_Williams (~jas_willi@217.41.234.60) |
| 06:30.55 | lehel | how do you do today? ladies and gentleman |
| 06:30.56 | *** join/#asterisk pchenry (~pchenry@firewall1.enetonline.net) |
| 06:31.06 | pchenry | anyone here this late ? |
| 06:31.19 | lehel | late?.. not everywhere |
| 06:31.19 | opus_ | These voip provider don't know a damn thing about mission critical servers. |
| 06:31.28 | Jas_Williams | MOrning 7:31 here not late at all :) |
| 06:31.38 | pchenry | is broadvoice having trouble ??? or does anyone here use them ? |
| 06:31.41 | lehel | Morning 9:31 ! |
| 06:32.04 | lehel | gentleman!.. my Asterisk is running! BUT: |
| 06:32.10 | opus_ | pchenry - yup. can you get in to 9784187300 (broadvoice support) |
| 06:32.11 | opus_ | ? |
| 06:32.21 | lehel | i have no dial tone |
| 06:32.41 | pchenry | opus ... is that a YES to they are down ? |
| 06:32.51 | Jas_Williams | lehel, on what type of device ? |
| 06:32.52 | opus_ | yes |
| 06:32.56 | pchenry | ahhhh |
| 06:33.04 | opus_ | i guess |
| 06:33.10 | opus_ | Got SIP response 404 "Not Found" back from 147.135.0.128 |
| 06:33.10 | opus_ | Urgent handler |
| 06:33.11 | pchenry | I know the CTO.... Ill call him |
| 06:33.13 | lehel | Wildcard TDM400 X100P |
| 06:33.20 | lehel | 4 fxs |
| 06:33.28 | opus_ | pchenry - could you, thanks. |
| 06:33.38 | Jas_Williams | lehel, two cards ? |
| 06:33.42 | lehel | Asterisk Ready. |
| 06:33.42 | lehel | *CLI> |
| 06:34.00 | opus_ | pchenry - ask him if he needs anybody to watch his servers , tell him 'opus' on #asterisk wants a job |
| 06:34.12 | opus_ | :) |
| 06:34.26 | Jas_Williams | lehel, post the output of ztcfg -vvvv to pastebin.ca |
| 06:35.06 | pchenry | he isnt answering his cell ..... must be workin on it |
| 06:35.12 | pchenry | opus were U able to get through to support |
| 06:35.21 | lehel | Jas_Williams: |
| 06:35.24 | lehel | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 06:35.24 | lehel | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 06:35.24 | lehel | Channel 03: FXO Kewlstart (Default) (Slaves: 03) |
| 06:35.25 | lehel | Channel 04: FXO Kewlstart (Default) (Slaves: 04) |
| 06:35.25 | lehel | 4 channels configured. |
| 06:35.26 | opus_ | pchenry no |
| 06:35.46 | *** join/#asterisk s9510278 (~bigdaddy2@203.210.152.35) |
| 06:36.01 | Jas_Williams | lehel, no X100p just a tdm400 |
| 06:36.04 | Jas_Williams | ? |
| 06:36.26 | pchenry | i guess they use their own system for internal lines |
| 06:36.28 | pchenry | stupid |
| 06:36.29 | opus_ | haha |
| 06:36.30 | s9510278 | hi, anyone experience Queue drop out after 90 secs during MOH? |
| 06:36.37 | opus_ | maybe they should use voicepulse |
| 06:36.45 | pchenry | something |
| 06:36.47 | *** join/#asterisk shidan (~shidan@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
| 06:37.17 | *** join/#asterisk Qorky (~Pooa@dsl-202-72-146-104.wa.westnet.com.au) |
| 06:37.34 | lehel | http://store.yahoo.com/asteriskpbx/witd4pofxsbu.html |
| 06:37.40 | shidan | anyone here set up ast_fax with exim? I switched from qmail last week |
| 06:37.46 | Jas_Williams | lehel, post your zapata.conf to pastebin.ca don't flood the channel |
| 06:38.08 | lehel | ok |
| 06:38.27 | s9510278 | HELP!!! 2 probs |
| 06:38.39 | *** join/#asterisk odie_flocon_ (~chatzilla@S01060011953d7c4c.cg.shawcable.net) |
| 06:39.17 | s9510278 | Prob 1. Calls suddently drop out of queue after approx. 90 secs while listening to music |
| 06:39.18 | Jas_Williams | s9510278, make sure you have an answer() somewhere in your extensions.conf before you queue the call |
| 06:39.52 | s9510278 | Jas_, I got the Answer b4 the Queue cmd |
| 06:39.57 | Jas_Williams | k |
| 06:40.16 | ManxPower | s9510278: make sure you don't have busydetect=yes or callprogress=yes |
| 06:40.35 | lehel | Jas_Williams: there it is |
| 06:40.42 | *** join/#asterisk apardo (~apardo@80-28-161-183.adsl.nuria.telefonica-data.net) |
| 06:41.09 | s9510278 | Prob 2. UniCall fails to send "Calling Party Category" if send_next_ANI signal is the same as send_calling_party_category. Any fixes for it? |
| 06:41.22 | pchenry | opus .... what do U use BV for ??? |
| 06:41.26 | pchenry | business or home ? |
| 06:41.56 | Jas_Williams | lehel, is that it ? you have no channels setup ? |
| 06:42.08 | lehel | i haven't? |
| 06:42.27 | lehel | i have!/.. at the end |
| 06:42.33 | opus_ | pchenry - Nothing because they go down so much I can't use them... |
| 06:42.43 | s9510278 | ManxPower, I'm using UniCall (not Zapata). busydectect and callprogress aren't there. |
| 06:42.45 | lehel | u see? |
| 06:42.53 | opus_ | pchenry - mostly for testing asterisk. my orig plan was to use it for business but they are unreliable. |
| 06:43.01 | pchenry | who do you use primarily ? |
| 06:43.13 | opus_ | my cellphone |
| 06:43.26 | pchenry | vonage seems to be alot more stable |
| 06:43.30 | pchenry | with asterisk at least |
| 06:43.31 | Jas_Williams | lehel, yes sorry, ok |
| 06:43.41 | opus_ | yeah, but, does it integrate with asterisk? |
| 06:43.47 | pchenry | yes |
| 06:43.54 | opus_ | didn't know that |
| 06:43.55 | Jas_Williams | lehel, what does zap show channels give ? |
| 06:44.04 | pchenry | U gotta buy a "softphone" account |
| 06:44.13 | pchenry | when did the 404 messages start for you |
| 06:45.13 | Qorky | whats the best place to keep updated on asterisk changes and features? and whats different versoin to version. I just want to know if im missing out on anything by not upgrading. |
| 06:45.15 | lehel | in the asterisk command line? |
| 06:45.16 | lehel | 0 active channel(s) |
| 06:45.16 | lehel | *CLI> |
| 06:45.35 | Jas_Williams | lehel, no zap show channels not show channels |
| 06:45.47 | opus_ | pchenry - as soon as I checked it. |
| 06:45.49 | kb1_kanobe | Qorky: subscribe to the asterisk-cvs mailing list |
| 06:45.54 | kb1_kanobe | or -dev. |
| 06:46.02 | opus_ | perhaps its been down for a day, a week, dunno |
| 06:46.09 | pchenry | no |
| 06:46.17 | pchenry | just went down about 2 hours ago |
| 06:46.17 | opus_ | did you get a hold of your friend? |
| 06:46.19 | kb1_kanobe | Qorky: however, all stable releases are feature-free for now. Only cvs-head is growing. |
| 06:46.22 | pchenry | no |
| 06:46.33 | opus_ | is his broadvoice account down as well? :) |
| 06:46.34 | Qorky | ok thanks. will check it out now. |
| 06:46.54 | pchenry | called his cell :-) |
| 06:46.56 | |Vulture| | opus_: I haven't had a problem until this install |
| 06:46.56 | lehel | Jas_Williams: |
| 06:46.56 | lehel | Chan Extension Context Language MusicOnHold |
| 06:46.56 | lehel | <PROTECTED> |
| 06:46.57 | lehel | <PROTECTED> |
| 06:47.23 | |Vulture| | opus_: 11 phones, 2 need RMA, one has the control buttons not working, and 1 has this on/off problem |
| 06:47.28 | Jas_Williams | lehel, looks good |
| 06:47.53 | ManxPower | Qorky: By not using CVS-HEAD you are missing out on significant changes on an almost weekly basis that sometimes break things. |
| 06:47.53 | Jas_Williams | lehel, have you connected the external power to the card ? |
| 06:48.30 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 06:48.52 | |Vulture| | opus_: I think I am just gunna get a few more from the place I keep buying them from and send 2 back say they are broken... RMA is sucky |
| 06:48.54 | lehel | i should'nt change the caller ID to asrecieved? |
| 06:48.58 | lehel | the external power? |
| 06:49.12 | opus_ | hmmm |
| 06:54.27 | *** join/#asterisk dasuberdavid (~david@pcp01931229pcs.huntsv01.al.comcast.net) |
| 06:58.01 | opus_ | vulture - you can power cycle them remotely somehow |
| 06:58.10 | opus_ | try mgcp instead of sip |
| 06:58.27 | lehel | i have dial-tone!!! asterisk!! |
| 06:58.31 | opus_ | yay! |
| 06:58.32 | opus_ | gnite |
| 06:58.49 | lehel | May 5 09:56:36 WARNING[16222]: chan_zap.c:1472 zt_set_hook: zt hook failed: Dev ice or resource busy |
| 06:58.49 | lehel | May 5 09:56:37 NOTICE[16811]: pbx.c:1330 pbx_extension_helper: Cannot find exte nsion context 'intern' |
| 06:59.29 | Jas_Williams | lehel, add a context [intern] to extensions.conf |
| 07:00.02 | lehel | where in the file? |
| 07:00.12 | Jas_Williams | at the end |
| 07:00.17 | Jas_Williams | add |
| 07:00.19 | lehel | i have [incoming] [outgoing].. after them? |
| 07:00.22 | lehel | ok |
| 07:00.28 | kb1_kanobe | grr.... I'm trying to get the fax detect set up with PRI. It detects and jumps to the fax exten just fine, but where do I find the DID information? Ie. we normally match the DID against the exten - is it stored somewhere after the jump to 'exten => fax'? |
| 07:00.58 | Jas_Williams | lehel, waht other phones do you have on the system ? |
| 07:01.31 | lehel | saved.. other phones? like? |
| 07:02.01 | Jas_Williams | SIP IAX external ? I what to allow you to call something |
| 07:03.26 | Jas_Williams | lehel, post your extensions.conf to pastebin and I can see what you have.... |
| 07:03.38 | lehel | k.. |
| 07:10.35 | Jas_Williams | lehel, what does your [outgoing] do ? |
| 07:10.51 | *** join/#asterisk Archi (Archimedes@pcp0010563592pcs.shklfd01.ar.comcast.net) |
| 07:10.55 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 07:11.01 | tengulre | hi,all |
| 07:11.11 | Archi | hello world |
| 07:11.28 | Jas_Williams | morning |
| 07:11.39 | Archi | man ton of people in here |
| 07:11.53 | tengulre | what is diffrent between asterisk and bayonne? |
| 07:13.19 | tengulre | afternoon |
| 07:13.21 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 07:14.09 | tengulre | pls |
| 07:14.22 | tengulre | answer me! |
| 07:14.46 | chaoscon | tengulre: relax and give people a chance to respond |
| 07:14.54 | Archi | morning... has anyone used asterisk to route outside calls to a VoIP Public Number, so to say Landline to a real number that is actually a VoIP user? |
| 07:16.10 | tengulre | hi,chaoscon,do you known "bayonne"? |
| 07:16.21 | chaoscon | tengulre: no |
| 07:16.50 | tengulre | chaoscon, oh |
| 07:17.03 | *** join/#asterisk jonathh1 (~asd@84.9.16.255) |
| 07:17.31 | tengulre | chaoscon,does the asterisk support ivr? |
| 07:18.34 | newl | tengulre: http://www.asterisk.org/index.php?menu=features |
| 07:19.09 | *** join/#asterisk james_ed (~james_ed@ip68-103-200-171.ks.ok.cox.net) |
| 07:20.11 | james_ed | asterisk implementation question for anyone who likes nubes |
| 07:20.31 | newl | Ask and ye may be answered. |
| 07:20.34 | james_ed | hehe |
| 07:20.36 | Archi | the reason I ask, is because I have a Company in germany that's interested in having american number route to they're VoIP phones in germany. |
| 07:21.10 | james_ed | well...i plan on implementing VoIP, however, i'm not that far yet |
| 07:21.15 | tengulre | chaoscon,how can i config the ivr flow? |
| 07:21.57 | newl | Archi: If you subscribe to a north american provider and have your asterisk box register to that provider, it shouldn't be a problem. |
| 07:22.29 | newl | tengulre: Have you visited the plethora of documentation on http://www.voip-info.org? |
| 07:23.13 | Archi | newl: see the whole register thing is what I'm missing.. how is this done, and what would I have to get from the Provider? a T1 or would a normal POTS Line do for starters? |
| 07:24.05 | newl | Archi: That depends on how many DID numbers you want configured both way or incoming only. |
| 07:24.38 | newl | Archi: alternatively you can go with a VoIP provider that can give you a DID number or range of numbers. |
| 07:25.29 | Archi | newl: let's say we start with 5 numbers both way's I would think since they want it to look local when they're Field Rep return calls. |
| 07:27.37 | tengulre | I want use asterisk in my Lan? when outside dial in the user's computer auto pop a screen! |
| 07:27.37 | tengulre | is it? |
| 07:28.12 | Archi | Tengulre: then your looking for a softphone sounds to me like. |
| 07:28.33 | newl | Archi: 5 DID's with a VoIP provider shouldn't be a hassle. The provider you choose would be able to provide you with the account details (and in some circumstances the basic asterisk configuration required). That being said, however, you will still have to configure Asterisk to properly handle the calls for those clients be it hardware or software based. |
| 07:29.06 | tengulre | Archi,yes , |
| 07:29.32 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 07:30.01 | newl | bah..need another gig of ram..this openembedded world build is really killing me |
| 07:30.04 | Qwell | to verify, DS1=T1? |
| 07:30.31 | Archi | newl: ok any provider in mind? what's the average cost for this? And if I didn't want to go though a provider but rather do it myself, in a sense becoming the provider? oh the questions. |
| 07:31.02 | Archi | tengulre: yes. |
| 07:32.18 | tengulre | Archi,I want develop client application myself,but i don't know how to do use asterisk popsrceen interface? |
| 07:32.26 | newl | Archi: I don't have any information regarding providers, so unfortunately I can't help you there. If you were to go the PRI route, there'd be alot more work involved. |
| 07:34.13 | Archi | but I could have theoretically one pot's line and multiple numbers on it and * just changes the CID according to who is placing the call through VoIP then though right? |
| 07:35.18 | Archi | tengulre: your going to have to be a bit more descriptive, define "client application".. you want to programm a softphone on your own? or build the PBX? |
| 07:37.34 | Archi | newl: do you have a good website that can explain the PRI DID part? |
| 07:37.35 | tengulre | Archi,I want to program a softphone myself, when outside call in the computer auto pop customer detail! |
| 07:37.51 | tengulre | as name,age,address ... |
| 07:38.12 | nDuff | tengulre, personally, I was looking at building a design to do that via sending XMPP messages from a proxy listening to the Asterisk management interface over to the client. |
| 07:38.54 | nDuff | tengulre, that way the proxy is the only thing to need privileged access to the asterisk server (and, particularly, you -don't- need to grant such access to the clients). |
| 07:39.13 | Archi | ten: ah gotcha, what I would do, is route the call to the rep's phone, then have the rep keep a web page open that refreshes with the info provided from * using the Caller ID. |
| 07:39.36 | tengulre | Archi,yes. |
| 07:39.45 | tengulre | nDuff,thank you |
| 07:39.47 | nDuff | tengulre, obviously, the proxy needs to know which phone numbers map to which Jabber addresses (so it can send appropriate events, and authenticate incoming requests) |
| 07:41.41 | nDuff | tengulre, unlike Archi's proposed design, mine doesn't require that a web browser be kept open -- instead, a native (or perhaps Java) application that stays hidden mostand pops up a window on receiving messages is the more obvious way to implement the client -- though in my case it would generally spawn a browser window anyhow (with our CRM software), this nonetheless allows it to work even when the user has no browser running, and prevent |
| 07:41.41 | nDuff | s any need for the user's system to do polling. |
| 07:42.27 | tengulre | nDuff,yes I sew. but I still don't know how begin? I want to use delphi tool in win32 platform. |
| 07:42.47 | Archi | shoot it's late.. I need to get some shut eye, I'll look this over some more and then deceide what I want to do. |
| 07:42.47 | nDuff | (just to clarify, XMPP is the protocol used by Jabber; and whereas Jabber is mostly used for IM, it's more generally applicable as a way to move XML messages between locations). |
| 07:44.02 | Archi | nDuff: yeah I agree your apprroach is more straighforward if you have the programming experience. |
| 07:44.17 | nDuff | tengulre, well, the proxy you'd be running on your asterisk box -- that's probably not win32, and you'd be better off using a language better suited to text processing like Python or Perl; both of those have good XMPP libraries available. |
| 07:44.36 | nDuff | tengulre, for the client, there seem to be Jabber interfaces to Delphi; google for them. |
| 07:45.53 | tengulre | nDuff,thank you very much! |
| 07:46.23 | tengulre | nDuff,are you programmer? |
| 07:46.44 | nDuff | tengulre, yes. (I'm rather too busy to accept new work, however). |
| 07:47.21 | Archi | nDuff: man you guy's are like accountants... those that have time are lousy and don't get the work done. |
| 07:47.30 | tzafrir | When exactly do you want asterisk to notify jabber? |
| 07:47.34 | tzafrir | you could always use the simple System(xmppsend whatever) in your dialplan |
| 07:47.38 | tzafrir | I haven't tried xmppsend, just saw it on my apt packages list |
| 07:47.48 | tengulre | nDuff,I'm too,but I program fax server before, don't know call center! |
| 07:48.00 | nDuff | Archi, heh. |
| 07:49.35 | nDuff | hmm, actually, this looks relevant: http://voip-info.org/wiki-Asterisk+Jabber |
| 07:50.58 | nDuff | a variation on that theme should at least be good enough to get the call notification out of Asterisk and into a Jabber-based app somewhere. O'Reilley has a book on doing jabber-related programming, btw... might be a useful thing if you were going to try to implement this project. |
| 07:51.06 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 07:52.26 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 07:54.48 | tzafrir | sorry, that sendxmpp and not xmppsend |
| 07:55.27 | Archi | well guy's thanks for the info, got some more reading todo.... in the morning see you guy's then. |
| 07:57.49 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-9-134.w82-124.abo.wanadoo.fr) |
| 08:00.51 | tengulre | hi, does the asterisk support fax function? |
| 08:01.13 | |Vulture| | yea with spandsp |
| 08:01.28 | |Vulture| | tengulre: well better yet... what exactly do you want * to do with fax? |
| 08:02.11 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:02.54 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 08:02.59 | PTG123 | anyone here familiar with using a sun? |
| 08:03.33 | Qwell | barely |
| 08:03.38 | PTG123 | um |
| 08:03.42 | PTG123 | mine won't power on |
| 08:03.46 | PTG123 | then i let it sit there for a while |
| 08:03.49 | PTG123 | something came up on screen |
| 08:03.53 | PTG123 | then went back to a blank screen |
| 08:03.54 | Qwell | what kind? |
| 08:03.56 | PTG123 | is there some trick |
| 08:03.57 | PTG123 | ultra 5 |
| 08:04.26 | Qwell | is it getting to the OBP (open boot prompt)? |
| 08:04.28 | PTG123 | would it not turn on if it didn't have a mouse? |
| 08:04.33 | PTG123 | nothing on screen at all |
| 08:04.37 | Qwell | shouldn't. keyboard however, no |
| 08:04.45 | PTG123 | it has a keyboard |
| 08:04.47 | Qwell | erm, should? damn double negatives |
| 08:04.48 | PTG123 | when i plug it in with it on |
| 08:04.52 | PTG123 | it beeps and lights flash on keyboard |
| 08:05.09 | Qwell | I can't say I've ever used a sun with a keyboard/monitor |
| 08:05.25 | *** join/#asterisk christo (~chris@212.18.226.160) |
| 08:05.27 | Qwell | I just know when I turn mine on, I hit the obp, and can do stuff from there |
| 08:05.36 | Qwell | its been quite a while since I've seen it though |
| 08:05.41 | christo | dead wiki? |
| 08:05.47 | PTG123 | do you have to push anything on keyboard to make it turn on |
| 08:05.50 | PTG123 | or something stupid like that? |
| 08:06.04 | Qwell | wouldn't know... |
| 08:06.06 | Qwell | try stop-a |
| 08:06.46 | Qwell | I think that will always take you to obp |
| 08:06.46 | PTG123 | nope |
| 08:06.50 | PTG123 | man i just bought a new motherboard to fix this |
| 08:06.53 | PTG123 | now i can't make it power up :) |
| 08:07.10 | Qwell | What comes up on screen? |
| 08:07.17 | PTG123 | nothing now power to monitor |
| 08:07.22 | PTG123 | er no power |
| 08:07.35 | Qwell | You said something came up on the screen, then went back to blank |
| 08:07.54 | PTG123 | yep |
| 08:07.55 | PTG123 | it was weird |
| 08:08.30 | Qwell | just like split second flash of text or something? |
| 08:08.33 | PTG123 | yes |
| 08:09.23 | Qwell | its a mostly Linux mailing list, but they help with general problems sometimes too. try the aurora-sparc-users list |
| 08:09.39 | Qwell | I've seen them fix quite a few general problems |
| 08:10.02 | PTG123 | man i hate computers |
| 08:10.06 | PTG123 | never seem to work when you need them to |
| 08:10.07 | Qwell | don't we all... |
| 08:10.54 | PTG123 | i think it could be keyboard |
| 08:10.57 | PTG123 | it needs some button to work |
| 08:11.10 | Qwell | stop-a is the only key combo I know, heh |
| 08:11.30 | PTG123 | well it has a power key on keyboard |
| 08:11.33 | PTG123 | maybe without it it won't power up |
| 08:11.45 | Qwell | probably more of a power down key |
| 08:11.50 | *** part/#asterisk mozrat (~mozrat@80.68.89.215) |
| 08:11.54 | *** join/#asterisk jonathh1 (~asd@host-84-9-19-183.bulldogdsl.com) |
| 08:12.05 | *** join/#asterisk kenshinblade (~maccheron@host37-9.pool80105.interbusiness.it) |
| 08:12.09 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 08:12.19 | lehel | hello people! |
| 08:12.20 | jonathh1 | hello people, how do i tell asterisk NOT to answer the incoming call on POTS using an x100p |
| 08:12.26 | Qwell | jonathh1: unplug it |
| 08:12.30 | jonathh1 | lol |
| 08:12.32 | jonathh1 | cunning |
| 08:12.34 | Qwell | a pbx is a pbx |
| 08:12.44 | jonathh1 | anything alittle less brute force? |
| 08:12.45 | Qwell | a pbx picks up all calls |
| 08:12.52 | PTG123 | was able to read the screen when it did it again |
| 08:12.54 | PTG123 | keyboard not present |
| 08:12.58 | Qwell | PTG123: hmm |
| 08:13.12 | Qwell | wrong port? |
| 08:13.22 | Qwell | there are two, right? |
| 08:13.31 | PTG123 | no just one |
| 08:13.52 | kenshinblade | how can I set my channels to make them unregister? I mean: I want to have dynamic host sip registering, but when the phone register as the new user, it still receives calls directed to the old user |
| 08:14.03 | Qwell | oh, right...does the mouse plug into the keyboard? |
| 08:14.15 | tzafrir | jonathh1, don't have Answer() in your dialplan? |
| 08:14.18 | Qwell | erm, no, that was my apple |
| 08:14.22 | lehel | it's running asterisk! i have dial-tone! i need help to configure voicemail.conf & extensions.conf |
| 08:14.27 | PTG123 | yah it does |
| 08:14.32 | PTG123 | no you are right it does :) |
| 08:14.36 | Qwell | I have a sun kb/mouse, but...heh |
| 08:14.40 | jonathh1 | i dont have one.. the incoming call goes to its own context.. and that context is empty |
| 08:14.48 | Qwell | Does the kb cable detach from the kb? I know it did on my apple |
| 08:14.50 | Qwell | maybe its on the wrong side |
| 08:15.16 | lehel | somebody could help? |
| 08:15.30 | PTG123 | no it doesn't on this one |
| 08:15.45 | lehel | tzafrir: r u here? |
| 08:16.02 | tzafrir | I'm sometimes here. |
| 08:16.08 | PTG123 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=20328&item=5771343152&rd=1#ebayphotohosting |
| 08:16.16 | PTG123 | almost thinking about buying this |
| 08:16.34 | Qwell | Thats a good price if it stays down there |
| 08:16.38 | lehel | i need to have a voicepulse account? |
| 08:16.44 | PTG123 | 9 hours left and no bidder |
| 08:16.44 | Qwell | I paid that much for my SS5, heh |
| 08:17.23 | PTG123 | i am sure i could get it |
| 08:17.23 | PTG123 | whats an SS5? :) |
| 08:17.23 | Qwell | SparcStation |
| 08:17.23 | Qwell | my beloved router. :) |
| 08:17.23 | PTG123 | i don't know if i would actually use the thing |
| 08:17.23 | PTG123 | does asterisk compile on it? |
| 08:17.23 | Qwell | haha |
| 08:17.23 | Qwell | it'd take 6 weeks |
| 08:17.27 | PTG123 | to compile? :) |
| 08:17.28 | Qwell | mine is 110mhz |
| 08:17.30 | PTG123 | why is that? |
| 08:17.35 | PTG123 | heh |
| 08:17.36 | PTG123 | oh |
| 08:17.38 | PTG123 | well but does it :) |
| 08:17.38 | Qwell | I bet it would though. |
| 08:17.44 | PTG123 | for purify i would get this |
| 08:18.25 | Qwell | That one actually looks fairly clean. Mine is a bit yellow |
| 08:18.32 | Qwell | kb and mouse were pretty bad too iirc |
| 08:18.38 | james_ed | this might sound like a dumb question..here goes |
| 08:18.48 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 08:18.53 | Qwell | That one is missing a CD drive though, makes it look ugly on the outside |
| 08:19.13 | james_ed | how do i know what extension my analog phone is on (i have tdm400 with fxs and 3 fxos) |
| 08:19.13 | PTG123 | well i can put in a cd drive |
| 08:19.14 | PTG123 | heh |
| 08:19.20 | *** join/#asterisk csleong (~leongcs@t1.schenker.com.sg) |
| 08:19.23 | Qwell | the cd drive in mine rocks my world |
| 08:19.30 | Qwell | its so smooth and sturdy |
| 08:19.39 | kenshinblade | help |
| 08:19.44 | wiz8291 | anyone got a sec to help? |
| 08:19.45 | csleong | Hello everyone |
| 08:19.54 | wiz8291 | every call i make gets rejected by my telco... |
| 08:20.20 | Qwell | I should buy a new scsi HD, and sell this one I've got |
| 08:20.31 | csleong | Anyone here got the setup to use Intel 836 generic modem to work with Asterisk ? |
| 08:20.33 | Qwell | 10k rpm in a 110mhz sparc...sad |
| 08:20.46 | PTG123 | i got a drawer filled with them |
| 08:20.47 | PTG123 | :) |
| 08:20.57 | Qwell | its 10x slower then my ata100 drives in my desktop |
| 08:21.01 | PTG123 | i am sending file 3 cases of 10krpm fibre channel drives |
| 08:21.12 | csleong | Anyone here got the setup to use Intel 536 generic modem to work with Asterisk ? |
| 08:21.16 | Qwell | nice |
| 08:21.27 | PTG123 | http://www-306.ibm.com/software/awdtools/purifyplus/unix/ |
| 08:21.29 | PTG123 | i want this |
| 08:21.31 | PTG123 | but download doesn't work |
| 08:22.08 | lehel | how can i define mailboxes!? Comedian Mail ? |
| 08:22.24 | wiz8291 | lehel: voicemail.conf |
| 08:22.51 | lehel | and there?.. |
| 08:22.54 | wiz8291 | mailbox => password,User's Name,email address |
| 08:23.06 | lehel | [default] |
| 08:23.07 | lehel | 1234 => 4242,Example Mailbox,root@localhost |
| 08:23.30 | wiz8291 | so add one line per mailbox in that format |
| 08:23.32 | wiz8291 | simple |
| 08:24.00 | PTG123 | you know how much that software costs |
| 08:24.03 | PTG123 | just found the price |
| 08:24.05 | PTG123 | $5,495 |
| 08:24.13 | lehel | what passwd .. what User?.. who's? |
| 08:24.33 | PTG123 | RATL PURIFYPLUS FOR LINUX AND UNIX AUTHORIZED USER LIC+SW MAINT 12 MO (D52ZYLL) |
| 08:25.20 | lehel | how can i call simply a local number? |
| 08:25.56 | christo | What is this warning trying to tell me? May 5 09:23:49 WARNING[14654]: app_addon_sql_mysql.c:318 aMYSQL_fetch: ast_MYSQL_fetch: numFields=1 |
| 08:27.54 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 08:28.46 | Qwell | off to bed |
| 08:28.52 | langals | Hi there....I am using meetme....I want to know if my clients are using GSM, then is the audio transcoded into ulaw and then back to gsm....or would meetme pass the gsm through? |
| 08:28.58 | wiz8291 | anyone got any info on PRI? when i dial i get a congested tone and no number will go through |
| 08:29.29 | Qwell | PTG123: good luck on getting that booted...you're in for a fun ride |
| 08:29.39 | Qwell | You can run BSD and Linux on there btw... |
| 08:30.10 | Qwell | I did a "boot net" install on mine...was sweet. dchp, tftp, nfs |
| 08:30.21 | PTG123 | hehe |
| 08:30.28 | Qwell | only took 2 full days to install... |
| 08:30.29 | PTG123 | whats the point i got pcs for that :)i want solaris |
| 08:30.34 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-171-165.arcor-ip.net) |
| 08:30.44 | Qwell | Solaris on i386? |
| 08:31.42 | Blackvel | who noticed a better quality with eyebeam instead of x-lite? and if you are using eyebeam, what headset do you use? my soundcard always produces an echo. can I only fix it with bluetooth? :) |
| 08:34.13 | PTG123 | usb headsets only with softphones |
| 08:34.21 | PTG123 | nah need a real sun if i want to run solaris :) |
| 08:35.48 | Blackvel | PTG123: this echo depends very much on my soundcard I use? hm? |
| 08:36.40 | Blackvel | this is weird. I know someone from UK who has xlite + BT headset and whenever I try to call him with my ZAP, this sound is of that really bad voice quality. this is not normal? |
| 08:38.56 | PTG123 | i just would never use anything but a usb headset |
| 08:39.00 | PTG123 | if you want reliable softphone experience |
| 08:39.54 | wiz8291 | how can i see which number asterisk is sending out accross the PRI? |
| 08:39.56 | Blackvel | sure |
| 08:40.01 | Blackvel | my experience is really bad so far |
| 08:40.20 | Blackvel | would not recommend to anyone :) |
| 08:40.52 | wiz8291 | anyone? |
| 08:43.05 | Blackvel | wiz8291: not me, but you have 21 channels? how many phones do you have? |
| 08:43.57 | wiz8291 | we have 31 channels |
| 08:44.02 | wiz8291 | 24 extensions |
| 08:44.35 | Blackvel | oh, this is E1 even? |
| 08:44.50 | Blackvel | not bad |
| 08:44.58 | Blackvel | it works? :) |
| 08:45.12 | wiz8291 | yup |
| 08:45.17 | wiz8291 | incoming works |
| 08:46.22 | lehel | *CLI> May 5 11:46:14 WARNING[17588]: app_voicemail.c:3356 vm_execmain: Couldn't read username |
| 08:47.33 | wiz8291 | Blackvel: i can't make any outgoing calls |
| 08:48.28 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 08:51.14 | Blackvel | lehel: finally something works on your sideß :) |
| 08:51.27 | Blackvel | wiz8291: if I had an idea I would love to help you |
| 08:52.37 | lehel | Blackvel: could you help me pls to configure the voicemail.conf? pls |
| 08:52.49 | lehel | or the extensions.conf.. |
| 08:53.01 | wiz8291 | lehel: what are you trying to do? |
| 08:53.17 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 08:53.20 | lehel | to call |
| 08:53.35 | Blackvel | lehel: to call you dont need voicemail :) |
| 08:53.43 | lehel | ;) when i'm calling "1000" asks me a mailbox and passwd |
| 08:53.52 | lehel | then? |
| 08:54.03 | wiz8291 | then enter the mailbox number from the voicemail.conf |
| 08:54.05 | Blackvel | there is a good example in voicemail |
| 08:54.09 | wiz8291 | as we said, one line per mailbox |
| 08:54.13 | wiz8291 | its very very simple |
| 08:54.21 | Blackvel | the pwd is in voicemail lehel |
| 08:54.27 | lehel | not for me:(.. |
| 08:54.40 | lehel | [default] |
| 08:54.41 | lehel | 1234 => 4242,Example Mailbox,root@localhost |
| 08:54.48 | lehel | this is in the voicemail.conf |
| 08:55.19 | lehel | even i don't know what is that "mailbox" in the asterisk world |
| 08:55.35 | Blackvel | pwd is 4242 then |
| 08:55.43 | Blackvel | and voicemail number 1234 |
| 08:55.54 | Blackvel | whats that difficult with that? :) |
| 08:58.22 | lehel | and in the extensions.conf am i adding anything? |
| 08:58.30 | wiz8291 | lehel: no |
| 08:58.38 | lehel | k |
| 08:59.18 | lehel | 2000 => 2001,lehel,lehel@xxx.x < ok? |
| 08:59.55 | wiz8291 | yeah |
| 09:00.04 | wiz8291 | then your mailbox is 2000 and your pwd is 2001 |
| 09:00.24 | lehel | CLI> May 5 12:00:13 WARNING[17613]: app_voicemail.c:3356 vm_execmain: Couldn't read username |
| 09:00.29 | lehel | ? |
| 09:01.09 | lehel | wait!.. let's start at the beginning! |
| 09:01.13 | lehel | my asterisk is running |
| 09:01.16 | lehel | i have dial-tone |
| 09:01.30 | lehel | now what is the next step? |
| 09:02.32 | Blackvel | lehel: dont touch voicemail for outgoing calls |
| 09:04.19 | *** part/#asterisk csleong (~leongcs@t1.schenker.com.sg) |
| 09:04.43 | lehel | how to call out? |
| 09:06.09 | christo | I'm experimenting with the Monitor command. I play a series of beeps over the phone to the user (using the Background command) and I just say 'beep beep' in time with the beeps. The problem is that when they are mixed together, my spoken 'beeps' are fractionally lagged. What could be causing this? |
| 09:07.01 | Blackvel | lehel: with what? zap? pri? sip provider? |
| 09:07.08 | lehel | zap |
| 09:07.32 | Blackvel | lehel: extensions.conf exten => _91.,1,Dial(ZAP/g1/<number>,30,r) |
| 09:07.53 | Blackvel | make sure you have a group, otherwise use ZAP/ressource/number |
| 09:08.01 | Blackvel | tdm400p? |
| 09:08.07 | lehel | yes |
| 09:08.10 | Blackvel | probably you can use zap/1/<number> there |
| 09:08.15 | Blackvel | if 1 is fxo |
| 09:08.18 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 09:09.48 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:10.39 | Zeeek | heh |
| 09:11.56 | Zeeek | any nufone users? |
| 09:14.57 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 09:14.59 | *** join/#asterisk cjk_ (~cjk@80.92.75.4) |
| 09:17.07 | james_ed | anyone familiar with the rapid load? |
| 09:20.11 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 09:20.28 | james_ed | i'm trying to set up an autoattendant...is this basically setting up a new dial plan? |
| 09:29.19 | james_ed | if i have fxs module on tdm400p, how can i tell what extension my analog phone is connected to the fxs port? |
| 09:30.05 | Zeeek | what have you done so far? |
| 09:31.25 | Zeeek | your question is a little vague |
| 09:31.40 | james_ed | ok..sorry |
| 09:31.41 | Zeeek | nufone, anyone? I have a DID problem |
| 09:31.52 | james_ed | i installed xorcom rapid |
| 09:31.58 | Zeeek | ok |
| 09:32.00 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 09:32.03 | Zeeek | that explains everything |
| 09:32.06 | james_ed | haha |
| 09:32.20 | james_ed | i guess i'm not understanding how dial plan works |
| 09:32.22 | Zeeek | maybe tzafrir will be around in a while to help you |
| 09:32.36 | Zeeek | james_ed that's the big problem with all the easy setups |
| 09:32.46 | james_ed | understood |
| 09:32.50 | Zeeek | you could consider learning about dialplans |
| 09:32.52 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction |
| 09:32.52 | Zeeek | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 09:32.52 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Extension+Matching |
| 09:32.52 | Zeeek | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 09:32.57 | james_ed | i'm reading right now...thx |
| 09:33.20 | james_ed | what i really need to do is set up a menu (autoattendant) |
| 09:33.21 | Zeeek | long story short - your FXS will appear as a ZAP channel |
| 09:33.25 | james_ed | ok |
| 09:33.47 | Zeeek | then you add statements to extensions that make an extansion number call the ZAP channel |
| 09:34.17 | james_ed | i guess that makes sense |
| 09:34.28 | Zeeek | if you want to know, read up |
| 09:34.35 | james_ed | thx |
| 09:34.44 | Zeeek | there are auto att examples in almost every doc |
| 09:34.45 | Zeeek | Starter tutorial: |
| 09:34.45 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:34.45 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:34.45 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 09:34.45 | Zeeek | THE reference of the moment: |
| 09:34.46 | Zeeek | http://www.asteriskdocs.org |
| 09:34.54 | james_ed | sweet |
| 09:35.08 | james_ed | i have 3 pstn lines...3 fxos on my tdm400 |
| 09:35.11 | Zeeek | I badly need someone to help me with nufone DID codec problem before I leave the country |
| 09:35.22 | wiz8291 | i have ISDN30e, but can't make any calls :( |
| 09:36.26 | james_ed | i have a stand alone office jet for faxing...can i split my phone cord going into fxs port to share with fax and cordless? |
| 09:37.00 | james_ed | hmm maybe that wouldn't work very well |
| 09:38.54 | james_ed | well..ok all my phone lines are going into tdm400 P...how can i use my office jet 7410 for faxing? |
| 09:39.31 | langals | Hi there....if I am using meetme, and my clients are using gsm...then is meetme transcoding from gsm to ulaw and back to gsm, or does it pass gsm through? |
| 09:49.55 | *** join/#asterisk tomassia (~user@203.131.164.162) |
| 09:50.25 | wiz8291 | hmmmm |
| 09:50.26 | wiz8291 | ok |
| 09:50.28 | wiz8291 | this is weird |
| 09:50.48 | wiz8291 | when i dial 7733281590 to the pstn it works, if i dial it normally (with the leading 0) it does not work... |
| 09:50.56 | wiz8291 | any ideas? |
| 09:52.38 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:55.59 | fenlander | wiz8291: what is your dialplan like? |
| 09:56.00 | *** part/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 09:57.09 | Zeeek | anyone with nufone DID here? |
| 09:57.16 | kb1_kanobe | wiz8291: pri? |
| 09:57.26 | Blackvel | who uses eyebeam/xlite with headset? |
| 09:57.42 | Zeeek | used to use x-lite |
| 09:58.12 | kb1_kanobe | wiz8291: you've got the wrong switchtype configured for your carrier. Try 'unknown' and see if that helps. |
| 09:58.14 | kb1_kanobe | gnight |
| 09:58.39 | Weezey | Zeeek: I heard last night that they were having some issues. |
| 09:58.51 | Zeeek | with codec 0x800 ? |
| 09:58.56 | Weezey | Blackvel: one of my customers does all the time. |
| 09:59.10 | Weezey | Zeeek: I know no details. |
| 09:59.34 | Zeeek | link to where you heard? mailing list? |
| 09:59.47 | Zeeek | Paris Hilton on Leno ? |
| 10:00.00 | tomassia | can anybody tell me what is rtp codec 72 |
| 10:00.25 | facek_ | anyone use linksys pap2 phone adapter? |
| 10:00.35 | Zeeek | ok now I see more normal behavior. Is it possible to record sound files in g729? |
| 10:01.21 | Zeeek | I'm getting Unable to find a path from gsm to g729 when I try to Background a file |
| 10:01.46 | wiz8291 | switchtype is not wrong though |
| 10:01.50 | wiz8291 | its euroisdn :( |
| 10:01.51 | Weezey | Zeeek: I'm told you can. |
| 10:02.04 | Zeeek | let me look at sox |
| 10:03.36 | Weezey | Zeeek: http://lists.digium.com/pipermail/asterisk-users/2003-October/024655.html |
| 10:04.50 | s9510278 | can anyone explains how queue/agents/Local channels/etc. works together, i mean in terms of which channels invoked by which channels and in what order? thanks! |
| 10:08.43 | Zeeek | Weezey thanks but file.c:224 ast_writestream: Unable to translate to format g729, source format ulaw |
| 10:09.04 | Zeeek | <PROTECTED> |
| 10:09.06 | tzafrir | james_ed, what's up? |
| 10:09.08 | Weezey | I thought caller ended up in queue, next available agent rings unless they're all full, at which point caller stays in queue. As for Local channel, I never knew about it, but it basically allows you call into your dialplan. |
| 10:10.12 | Weezey | Zeeek: maybe you need a license to create the g729 files? |
| 10:10.19 | Zeeek | I have 4 licenses |
| 10:10.29 | Zeeek | there is some weirdness |
| 10:10.35 | Weezey | hrmmm, then that's not helping. |
| 10:10.37 | Zeeek | I can't seem to call the DID |
| 10:10.47 | Zeeek | and playback files |
| 10:11.06 | Weezey | is the call coming in g729? |
| 10:11.19 | Zeeek | no, gsm I think |
| 10:11.41 | Weezey | really? That's cool. |
| 10:11.56 | Weezey | but doesn't explain your problem. |
| 10:12.05 | Zeeek | when the call is answered, here's what happens |
| 10:12.07 | Zeeek | channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729 |
| 10:12.22 | Zeeek | <PROTECTED> |
| 10:12.55 | Zeeek | the record g729 idea was because of this: |
| 10:12.58 | Zeeek | Unable to open rr/vacation (format g729): No such file or directory |
| 10:13.07 | cjk_ | anyone here who can help me with txfax? rxfax is working greate, but txfax only sends me white pages |
| 10:13.10 | Zeeek | it tried to open a g729 version for some reason |
| 10:13.22 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-47.dsl.scarlet.be) |
| 10:13.33 | Zeeek | maybe if the file existed in wav? |
| 10:13.37 | Zeeek | easy enuf to try |
| 10:13.53 | Zeeek | wav is ulaw, right? |
| 10:15.02 | Zeeek | anyone using Nufone - can you confirm the codec list to use with them now? |
| 10:17.48 | onkeltimm | I just upgraded to CVS_HEAD, and now cannot get chan_capi to compile or load anymore... any insights? |
| 10:20.44 | onkeltimm | here' s the compile output: http://pastebin.ca/11019 |
| 10:21.14 | Blackvel | zeeek: had you any problems with echo? |
| 10:21.33 | Zeeek | there is always a little echo |
| 10:21.42 | Zeeek | in the beginning of calls, anyway |
| 10:21.45 | Blackvel | weezey: ah, and he uses bluetooth headsets? |
| 10:22.01 | Blackvel | hm |
| 10:22.13 | Blackvel | weird. does not happen with ZAP hardware :) |
| 10:22.33 | Blackvel | my soundcard always produces echo |
| 10:22.38 | Blackvel | maybe wrong card |
| 10:22.40 | Zeeek | ok, I removed the playback just to try DISA directly. I answer() do DISA and still it wants g729 |
| 10:23.03 | Blackvel | disa has nothing to do with g729 :) |
| 10:23.03 | onkeltimm | is chan_capi not working with CVS_HEAD? |
| 10:23.39 | Zeeek | I mention DISA because there are no longer sound files involved |
| 10:23.41 | *** join/#asterisk UK_Mister (~fred@host81-137-167-89.in-addr.btopenworld.com) |
| 10:23.54 | UK_Mister | hello |
| 10:24.01 | Zeeek | after the DISA line: channel.c:1691 ast_set_write_format: Unable to find a path from ulaw to g729 |
| 10:24.14 | UK_Mister | anyone used a Grandstream GXP-2000?? |
| 10:24.15 | onkeltimm | also, when I start * it just gives me "Ouch ... error while writing audio data: : Broken pipe" |
| 10:24.16 | Zeeek | why does it look for g729? |
| 10:24.44 | Zeeek | onkeltimm I think you need to noload= a module like alsa |
| 10:25.08 | tomassia | can anyone tell me how to fix the unknown RTP codec 72 problem? thanks |
| 10:25.13 | james_ed | tzafrir...where can i find info about rapids config files and how to customize them? |
| 10:25.17 | Blackvel | zeeek: you need to have a g729 license for this then |
| 10:25.17 | Zeeek | Blackvel I find echo is very dependent on the level of audio |
| 10:25.34 | Zeeek | I do have 4 licenses |
| 10:25.36 | Blackvel | oh |
| 10:25.39 | Zeeek | and g729 works |
| 10:25.39 | tzafrir | james_ed, basically standard Asterisk configurations |
| 10:25.46 | Blackvel | are yuo sure your g729 module is loaded? |
| 10:25.47 | onkeltimm | Zeeek: I have that noload => chan_alsa.so |
| 10:25.57 | Blackvel | how do you call into system? |
| 10:25.57 | tzafrir | so ~docs that was mentioned earlier applies |
| 10:26.23 | Zeeek | onkeltimm what about chan_oss ? |
| 10:26.44 | tzafrir | james_ed, take a look at subdirectories of /etc/asterisk/ |
| 10:26.57 | Zeeek | Blackvel ya if I force the use of g729 on say the sip phone, it works fine |
| 10:26.57 | james_ed | that's where i'm at right now |
| 10:27.13 | onkeltimm | Zeeek: noload line was commented out. activated it, but same error |
| 10:27.19 | Blackvel | zeeek: and ulaw-ulaw does not work? |
| 10:27.33 | tzafrir | james_ed, basically the files in .d directory are #include-d as-is in a certain config file |
| 10:27.37 | Zeeek | sure it does. This behavior is only with nufone DID |
| 10:27.44 | Blackvel | how does your sip.conf look like? disallow=all? allow=? what are the first codecs? |
| 10:27.49 | Zeeek | it's iax |
| 10:27.50 | james_ed | tzafrir...do u know if cisco 7910 can handle SIP? |
| 10:28.06 | Blackvel | zeeek: maybe nufone has problem with codecs |
| 10:28.09 | onkeltimm | and trying to load chan_capi * tells me loader.c:306 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed |
| 10:28.11 | UK_Mister | anyone got a view on the Grandstream GXP-2000 handset, anyone used one? |
| 10:28.12 | tzafrir | sorry, I have noidea |
| 10:28.13 | Blackvel | not all providers support all codecs |
| 10:28.36 | Blackvel | and only if the provider supports the codec, you are allowed to choose the best (between caller and your *) |
| 10:29.00 | james_ed | ...so if i want autoattendant...that is just basically setting up a dial plan with menus?..it's basically voicemail? |
| 10:30.07 | tzafrir | a simple autoatendant is basically playing a sound file and then sending the user to a certain context (technically not in that order) |
| 10:30.32 | tzafrir | james_ed, apt-get install asterisk-docs |
| 10:30.42 | tzafrir | The original configuration examples are there |
| 10:30.48 | james_ed | ok |
| 10:30.53 | Blackvel | zeeek: have you checked iax.conf yet? |
| 10:30.59 | tzafrir | under /usr/share/doc/asterisk-doc/examples |
| 10:31.22 | Zeeek | Blackvel yes, I've been playing with that for an hour |
| 10:31.24 | james_ed | ...do i create sound files...?mp3s? |
| 10:31.31 | Zeeek | I know nufone supports gsm, ilbc |
| 10:31.47 | tzafrir | james_ed, you could use wav (or gsm) |
| 10:32.06 | tzafrir | There are some slight technical problems with mp3 ATM |
| 10:32.22 | dpryo | Isn't wav just a container format? What codec do you mean, tzafrir? |
| 10:32.45 | *** join/#asterisk meshugga (philip@loeblich.linuxteam.at) |
| 10:32.47 | dpryo | I mean.. you can easily encode mp3s with wav-headers ;) |
| 10:32.49 | meshugga | hi |
| 10:32.50 | Blackvel | zeeek: maybe the problem is not on your side (had previous experiences with sipgate, nikotel and puretel) |
| 10:32.55 | tzafrir | dpryo, hmm, well, the standard ones supported by the default installation |
| 10:33.00 | meshugga | i need some documentation for the chan api |
| 10:33.04 | dpryo | tzafrir: Pure pcm perhaps ;) |
| 10:33.21 | dpryo | The most common 'wav' |
| 10:33.22 | meshugga | or some very good coded chan module which implements "incoming" calls |
| 10:33.30 | Blackvel | meshugga: there is none :) |
| 10:33.34 | tzafrir | wav isn't pure pcm. It has at least a header so you'd know what to expect |
| 10:33.46 | dpryo | tzafrir: Yeah, but inside the container. |
| 10:34.07 | meshugga | ok, so how about anyone who has experience with channel module coding/debugging? |
| 10:34.18 | james_ed | is capturing audio for messages in the docs? |
| 10:35.56 | Blackvel | meshugga: you can debug chan_capi I think |
| 10:36.03 | Blackvel | with bristuff it works |
| 10:36.50 | meshugga | can you name the function which "takes" the call and forwards it to asterisk? |
| 10:37.07 | Blackvel | not me |
| 10:39.23 | *** join/#asterisk syle (~blah@wnpgmb02dc1-58-35.dynamic.mts.net) |
| 10:41.56 | onkeltimm | hm I downgraded to 1.0.7 and now it fscking segfaults! |
| 10:42.07 | onkeltimm | then it gives me "Ouch ... error while writing audio data: : Broken pipe" |
| 10:42.18 | onkeltimm | and "Warning, flexibel rate not heavily tested!" |
| 10:42.29 | onkeltimm | please help... |
| 10:44.30 | kajtzu | blah. latest cvs doesn't compile on x86_64 |
| 10:45.09 | onkeltimm | how can I get rid of that error while writing audio data? |
| 10:49.31 | onkeltimm | i have loads mpg123 processes left running... could it be something with that? |
| 10:52.28 | james_ed | how does asterisk handle incoming fax? |
| 10:53.37 | onkeltimm | ok found out that the problem is the segmentation fault * gives me. what misconfiguration can cause * to segfault? |
| 10:56.16 | onkeltimm | it segfaults after loading chan_features.so ... and after that I can see my IAX registrations... weird |
| 10:57.54 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 11:00.52 | tzafrir | james_ed, technically you could use rxfax |
| 11:03.13 | Blackvel | onkeltimm: you downgraded to 1.0.7? ah you mean from cvs head version? |
| 11:03.58 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 11:04.04 | onkeltimm | yeah, first i upgraded from 1.0.3 and after that * segfaults on me and wont load chan_capi. then I did the downgrade do 1.0.7 but still it segfaults, even with samplke config... |
| 11:04.20 | Sato1 | actually, incoming fax can be redirected to an extension or to an app, see the extensions section in voip-info.org to know how to add the "fax" option into your main menu |
| 11:05.29 | Sato1 | onkeltimm, do you use chan_capi? |
| 11:05.36 | james_ed | thx sato |
| 11:05.48 | onkeltimm | Sato1, yes i do. |
| 11:05.56 | Sato1 | james_ed, no problem |
| 11:06.58 | Sato1 | onkeltimm, that look like a chan_capi issue, not an asterisk issue, as chan_capi is not part of the asterisk |
| 11:07.01 | Blackvel | onkeltimm: you do something wrong |
| 11:07.07 | onkeltimm | strange is, with sampüle config it still loads chan_capi, I thought it wouldn't. do I have to remove it from modules directory? |
| 11:07.10 | Blackvel | you cant change asterisk and integrate chan_capi |
| 11:07.21 | onkeltimm | Blackvel: I thought so. |
| 11:07.25 | Blackvel | get chan_capi .35 version. it will download the correct asterisk source for you |
| 11:07.42 | Blackvel | I am really not wondering why it segfaults |
| 11:07.48 | Blackvel | codes do not match |
| 11:09.05 | onkeltimm | I have chan_capi .35 |
| 11:09.15 | onkeltimm | it compiles ok now with 1.0.7 |
| 11:09.27 | onkeltimm | how could it dl the correct sorce for me? |
| 11:12.03 | onkeltimm | i completely removed chan_capi. it still segfaults. |
| 11:12.18 | Blackvel | ./install.sh |
| 11:12.26 | Blackvel | sure |
| 11:12.29 | Blackvel | because it patches code |
| 11:12.34 | Blackvel | remove your complete asterisk |
| 11:12.45 | Blackvel | and do a fresh installation with install.sh |
| 11:14.35 | onkeltimm | my chan_capi source dir does not have any install.sh |
| 11:14.53 | onkeltimm | dled it anew, no install.sh |
| 11:15.28 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 11:15.30 | onkeltimm | which install.sh do you mean? |
| 11:16.47 | *** join/#asterisk dercol (~ercolani@sei.yacme.com) |
| 11:21.58 | *** join/#asterisk meppl (~mephisto@p54AACE0A.dip.t-dialin.net) |
| 11:22.20 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 11:22.27 | lehel | hello |
| 11:22.34 | meppl | guten morgen |
| 11:23.13 | meppl | good morning lehel |
| 11:23.38 | lehel | meppl: could you help me? |
| 11:23.46 | lehel | asterisk conf |
| 11:24.54 | meppl | emh, no |
| 11:24.55 | meppl | sorry |
| 11:25.36 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:26.00 | onkeltimm | Blackvel: i reverted to backup. it still segfaults... |
| 11:36.47 | Blackvel | backup? |
| 11:36.47 | Blackvel | throw away completely |
| 11:36.47 | Blackvel | start from scratch |
| 11:36.47 | Blackvel | maybe also a config problem |
| 11:37.11 | onkeltimm | how do i start from scratch? do you have a pointer to a doc on how to remove * completely? |
| 11:37.43 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 11:37.49 | Weezey | Happy 05/05/05 |
| 11:38.27 | Blackvel | yes |
| 11:38.45 | syle | wonderful |
| 11:38.52 | syle | much closer to june finally |
| 11:38.56 | Blackvel | rm -rf /usr/lib/asterisk |
| 11:39.02 | Blackvel | rm -rf /usr/src/asterisk |
| 11:39.10 | Blackvel | or rm -rf/usr/src/chancapi35 |
| 11:39.20 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 11:39.31 | Blackvel | you dont need to replace /usr/sbin/asterisk really |
| 11:39.38 | Blackvel | make install does automatically |
| 11:39.45 | Blackvel | only modules are from time to time a problem |
| 11:41.13 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 11:41.57 | Weezey | Blackvel: the latest CVS tells you when you install that there are old modules installed |
| 11:41.59 | Weezey | it's cool. |
| 11:42.42 | Blackvel | ah |
| 11:42.53 | Blackvel | can remember the times I had problems with different versions |
| 11:43.06 | Weezey | probably why someome spruced up the install |
| 11:43.48 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:44.19 | Blackvel | :) |
| 11:46.57 | key2 | what is congestion for ? |
| 11:48.43 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 11:48.56 | lehel | zzzzzzzz |
| 11:50.19 | Weezey | key2: so your lungs don't get filled with allergens |
| 11:50.28 | key2 | :) |
| 11:50.30 | key2 | why |
| 11:51.34 | Weezey | otherwise you'd die from breathing. |
| 11:52.52 | meshugga | how can a channel tell asterisk that it has an incoming call? |
| 11:52.54 | key2 | Weezey: what's Congestion() for |
| 11:52.55 | key2 | ? |
| 11:53.19 | Weezey | to play that swell noise meaning nothing worked. |
| 11:53.29 | meshugga | ast_setstate()? with AST_STATE_RING? |
| 11:55.17 | key2 | ok |
| 11:55.30 | key2 | Weezey: when I do an AGI in C for example, where do I leave the file ? |
| 11:56.25 | cypromis | /var/lib/asterisk/agi-bin |
| 11:56.33 | key2 | cypromis: should I compile it ? |
| 11:56.45 | key2 | let say I make an agi test.c |
| 11:57.01 | key2 | should I lieave just the bin test in /var/lib/asterisk/agi-bin |
| 11:57.02 | key2 | ? |
| 12:02.27 | Weezey | yes |
| 12:03.11 | Weezey | has anyone purchased licenses for g729 from digium? |
| 12:03.24 | Weezey | can I buy a few now and then add more later? |
| 12:05.48 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 12:06.03 | Zeeek | it's easier to buy them all at once, but yeah i did that |
| 12:08.24 | onkeltimm | Blackvel: ok I have 1.0.7 running now... with chan_capi... |
| 12:08.47 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:09.03 | onkeltimm | anyone can tell me what is the ABSOLUTE MINIMUM * version that has attended transfers integrated? |
| 12:09.48 | Weezey | Zeeek: okay, but it does work though? How is the sound quality with g729? Sound just like ulaw? |
| 12:10.57 | james_ed | how do u activate eth0 in command line? |
| 12:11.18 | Zeeek | Weezey it works ok. I'm not a big fan of g729 |
| 12:11.19 | Weezey | ifconfig eth0 up |
| 12:11.30 | james_ed | thx |
| 12:11.32 | Weezey | really? hrmm, maybe I shouldn't get it then. |
| 12:11.43 | Weezey | why aren't you a fan? |
| 12:11.55 | Zeeek | well i did to try it but i'm not that thrilled - we never have multiple calls so bw isn't a big issue |
| 12:12.16 | Weezey | So it doesn't sound all that great then? |
| 12:12.27 | Zeeek | needing a license is shitty. I don't mind paying, it's the license stuff tho |
| 12:12.56 | Weezey | I don't care about that, I want to know how it sounds. |
| 12:13.02 | james_ed | does ifconfig tell me ip addres? |
| 12:13.15 | Zeeek | you'll have to try it yourself for that :) |
| 12:13.18 | Weezey | did you try just typinging in ifconfig? |
| 12:13.24 | Zeeek | I bought one lic to do that |
| 12:13.41 | Weezey | and, does it sound the same? |
| 12:14.06 | Weezey | (I think I'd have to buy two by my count, one in, one out) |
| 12:14.19 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:15.45 | onkeltimm | no-one using attended transfers and chan_capi on one * box? |
| 12:17.18 | ManxPower | most people use the native transfer on the phone they use |
| 12:19.01 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:19.56 | onkeltimm | hm I have Atcom AT320ED IAX2-loaded phones here. work like a breeze but do not have a transfer button. |
| 12:20.12 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 12:25.43 | *** join/#asterisk agaran (agaran@agaran.pld-linux) |
| 12:25.50 | agaran | hello |
| 12:28.06 | Blackvel | asterisk integrates the feature attended transfers yet? |
| 12:28.33 | MikeJ[Laptop] | Blackvel, yes |
| 12:28.35 | Blackvel | onkeltimm: waht is your hardware? zap? I always thought (heard that months ago) that at least the zap drivers support attended transfers |
| 12:28.37 | MikeJ[Laptop] | ~docs |
| 12:28.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:29.00 | Blackvel | i wont search around in the docs |
| 12:29.00 | Blackvel | :) |
| 12:29.01 | MikeJ[Laptop] | search attended transfers |
| 12:29.07 | MikeJ[Laptop] | you won't |
| 12:29.10 | Blackvel | is there a bot command to search? |
| 12:29.15 | Blackvel | ~search attended transfers |
| 12:29.19 | onkeltimm | Blackvel: no zap. just the Atcom IAX phones and two fritzcards for pstn connectivity. |
| 12:29.19 | Blackvel | something like that? :) |
| 12:29.30 | Blackvel | atcom iax phones? |
| 12:29.31 | Blackvel | ah |
| 12:29.49 | Blackvel | some sip phones (snom e.g) have the transfer possiblity integrated too |
| 12:29.57 | MikeJ[Laptop] | you should still be able to use # transfers |
| 12:29.58 | Blackvel | had to write a queue on dec/jan 05 |
| 12:30.00 | Blackvel | quote |
| 12:30.20 | Blackvel | depends if asterisk integrates the code |
| 12:30.31 | onkeltimm | MikeJ[Laptop]: yeah, can do that, but these are blind. need to talk to the person i transfer to. |
| 12:30.37 | Blackvel | i didnt follow the bugid/patch |
| 12:30.55 | Blackvel | onkeltimm: check bugs.digium.com |
| 12:31.04 | Blackvel | there was the request to integrate that |
| 12:31.18 | Blackvel | I am not sure if that ever happend |
| 12:31.28 | onkeltimm | Blackvel: on the wiki I read that this is integrated. was in january or so. |
| 12:31.58 | MikeJ[Laptop] | you can use # for attended transfer |
| 12:32.15 | onkeltimm | ist here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20features.conf |
| 12:32.46 | onkeltimm | MikeJ[Laptop]: tried it, did not work. with my 1.0somethingcvs it did not |
| 12:33.17 | MikeJ[Laptop] | oh... not sure about stable |
| 12:33.24 | MikeJ[Laptop] | one sec |
| 12:33.46 | onkeltimm | and as I can't get chan_capi to cvompile with CVS_HEAD i have a problem there |
| 12:35.25 | james_ed | how the heck do i putty to my asterisk box...can't find the ip address on my local network |
| 12:35.31 | key2 | is there any way yo do a 3 way calling with asterisk using the ZAP ??? |
| 12:35.44 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:36.10 | onkeltimm | james_ed: man nmap ? ;) |
| 12:36.56 | MikeJ[Laptop] | attended #transfer is only in head |
| 12:37.22 | MikeJ[Laptop] | you can backport it maybe?? http://bugs.digium.com/view.php?id=3241 |
| 12:37.52 | MeTaBSD | Anyone use asterisk with Babytel here ? |
| 12:37.53 | key2 | so no one know if there is a way to do a 3 way calling with asterisk ? |
| 12:38.18 | MikeJ[Laptop] | like a flash type 3 way? |
| 12:38.26 | onkeltimm | MikeJ[Laptop]: maybe I could... but did no * devel work so far... let me get a login and I'll have a look |
| 12:38.48 | MikeJ[Laptop] | ummm, you will probably need to look at what was acutally commited. |
| 12:38.59 | MikeJ[Laptop] | i thought somone got capi working on head |
| 12:39.16 | MikeJ[Laptop] | ummm, maybe that was some other BRI stuff... |
| 12:40.48 | onkeltimm | don't understand a thing of the chan_capi code, but quite sure it's not running on head |
| 12:41.30 | Blackvel | you have to change it probably |
| 12:41.42 | MikeJ[Laptop] | I don't use it.. I don't know |
| 12:41.56 | Blackvel | depends watch patch tool tells you |
| 12:42.09 | Blackvel | if there are only minors, it is not a big problem |
| 12:42.16 | Blackvel | but it can get quite complex too |
| 12:42.21 | Blackvel | i know what I am speaking about |
| 12:42.21 | Blackvel | :) |
| 12:43.43 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 12:44.05 | MikeJ[Laptop] | do the cards support mISDN? |
| 12:44.34 | Blackvel | waht? |
| 12:44.50 | Blackvel | most 2.6 kernels support even passive isdn cards |
| 12:45.05 | Blackvel | best is to check with makeconfig |
| 12:45.33 | Blackvel | personally I dont know how good the chan_misdn integration with asterisk is |
| 12:45.39 | Blackvel | maybe not too bad |
| 12:45.40 | Aze` | How can i use attended xfer with Asterisk/SIP ? |
| 12:45.57 | Blackvel | cvs head? |
| 12:46.02 | wiz8291 | anyone got a second to help with group pickup? |
| 12:46.08 | MikeJ[Laptop] | it is in the works.. also, there is a head patch for can capi it appears on the wiki |
| 12:46.14 | onkeltimm | Blackvel: i need chan_capi because I use the two-fritzcards hack. |
| 12:46.32 | onkeltimm | running on 2.4 exactly for that reason |
| 12:46.32 | MikeJ[Laptop] | chan_capi.. |
| 12:46.47 | MikeJ[Laptop] | http://www.voip-info.org/wiki-Asterisk+CAPI+Channels |
| 12:47.42 | Aze` | i want only xfer features on 1.0.7.. i cant use head in production |
| 12:48.18 | wiz8291 | my pickup isn't working |
| 12:48.30 | MikeJ[Laptop] | ok, the link is up 10 minutes ago in this window. |
| 12:48.34 | wiz8291 | a phone is ringing, i dial *8 and i get the congestion tone |
| 12:48.38 | MikeJ[Laptop] | you will need to backport |
| 12:51.52 | ManxPower | Attended Transfer (using #) is NOT supported in 1.0.x |
| 12:52.06 | ManxPower | wiz8291: your SIP phone is eating the *8 |
| 12:52.37 | wiz8291 | this isn't a SIP phone |
| 12:52.39 | wiz8291 | all analogue |
| 12:53.06 | ManxPower | then make sure your callgroup and pickupgroup is listed BEFORE the channels in zaptel.conf |
| 12:53.27 | Aze` | ManxPawer > and in CVS head it's dont work well. How Can i resolve ? |
| 12:53.42 | wiz8291 | got that |
| 12:53.47 | wiz8291 | hmm, now it works |
| 12:53.48 | wiz8291 | this is odd |
| 12:54.08 | ManxPower | Aze`: You can work on fixing the code or pay someone else to fix the code. |
| 12:54.21 | wiz8291 | time to write up stuff on the wiki for E1 in the UK |
| 12:54.25 | ManxPower | Aze`: Or get a phone that supportes transfers |
| 12:55.30 | key2 | how is it possible to DIAL with an AGI ? |
| 12:55.32 | ManxPower | # transfers are only needed if you are using phones that are too stupid/cheap to support a transfer button |
| 12:55.52 | ManxPower | key2: read some of the billion AGI scripts out there that use Dial |
| 12:56.07 | key2 | cuz the fucntion wasnt in the agi doc :) |
| 12:56.18 | Aze` | ManxPawer i have phone that supportes transfers but it's very difficult to use by stupid receptionist |
| 12:56.42 | onkeltimm | Aze`: if you are willing to backport, i am with you. |
| 12:57.03 | ManxPower | key2: the AGI EXEC function is. |
| 12:57.23 | ManxPower | Asterisk 1.0.x will never get new features. |
| 12:57.29 | key2 | ManxPower: I wanted to do some 3 way conferencing with asterisk, basically I have to do a Flash Hook and then dial the number of the third person and then flash hook "3" to go into 3 way conferencing... how can I do that with asterisk ? |
| 12:57.41 | ManxPower | key2: Yes. |
| 12:57.42 | Blackvel | key2: send the EXEC DIAL message |
| 12:57.54 | ManxPower | Why not use meetme |
| 12:58.14 | key2 | meetme ? |
| 12:58.39 | Blackvel | ManxPower: sorry, didn't see you were faster ;) |
| 12:59.03 | MikeJ[Laptop] | Aze`. onkeltimm, the code did not look "too" invasive... it should not be a nightmare to backport |
| 12:59.59 | ManxPower | key2: perhaps you should become more familiar with Asterisk before trying to do complicated stuff. |
| 13:00.18 | key2 | ManxPower: is it already implemented ? |
| 13:01.09 | *** part/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 13:01.27 | ManxPower | key2: is what implimented? |
| 13:01.28 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 13:02.28 | *** join/#asterisk cmk (~cmk_@p54A3F09D.dip.t-dialin.net) |
| 13:03.59 | *** join/#asterisk t0pCop (t0pCop@221.128.98.158) |
| 13:04.18 | key2 | ManxPower: 3 way conferencing with zaptel |
| 13:04.42 | ManxPower | key2: it works just fine. |
| 13:04.48 | ManxPower | I do three-way calls all the time. |
| 13:04.55 | key2 | ManxPower: how do u do it |
| 13:05.11 | ManxPower | I press the FLASH button on my phone, dial the second number, then FLASH again. |
| 13:05.12 | key2 | using the same zaptel card |
| 13:05.42 | ManxPower | of course, if I want both legs of the call to go to the PSTN then I need two PSTN lines. |
| 13:05.53 | ManxPower | But that's just the way PBXs work. |
| 13:06.29 | key2 | ManxPower: my question is how can I call two PSTN numbers at the same time using my zap |
| 13:06.38 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 13:06.38 | ManxPower | key2: you need two PSTN lines. |
| 13:06.59 | key2 | ManxPower: in 3 way conferencing ? |
| 13:07.03 | ManxPower | key2: English is not your native language? |
| 13:07.11 | ManxPower | key2: yes. |
| 13:07.20 | key2 | well, hold on |
| 13:07.50 | key2 | from my PSTN line, with a normal phone, I can dial 2 numbers and pass into 3 way conferencing mode doing the FLASH thing |
| 13:07.55 | key2 | but how can I do that with asterisk |
| 13:08.17 | ManxPower | key2: Doing that with ONE PSTN line is a very complicated thing. |
| 13:08.40 | ManxPower | Asterisk was never designed to work with PSTN 3-way calling. |
| 13:08.51 | key2 | ok |
| 13:08.55 | key2 | I get it better :) |
| 13:09.00 | MikeJ[Laptop] | key2, the only way you can do that wil dialplan magic with flash sent to the phone line. look at the x100p call waiting stuff on the wiki, but it's not better |
| 13:09.28 | MikeJ[Laptop] | but it's not pretty that is |
| 13:10.24 | key2 | ManxPower: can I just dial a flash with asterisk using the ZAP ? |
| 13:10.51 | ManxPower | key2: no. |
| 13:11.04 | key2 | ManxPower: and what's the way to do it ? |
| 13:11.08 | MikeJ[Laptop] | there is a flash dialplan command |
| 13:11.25 | ManxPower | key2: the answer to your question would take me an hour to explain and I am not going to do that. |
| 13:11.32 | key2 | :) |
| 13:11.35 | MikeJ[Laptop] | but sending a flash from your phone is not somthing that you can trigger a flash on the line |
| 13:11.53 | *** part/#asterisk agaran (agaran@agaran.pld-linux) |
| 13:11.57 | key2 | MikeJ[Laptop]: whay u mean ? |
| 13:12.07 | MikeJ[Laptop] | because sending a flash from your phone is handled like the phone company handles it... it gives you another dialplan |
| 13:12.09 | key2 | with a voicemodem, when I dial the "!" it sends a flash |
| 13:12.12 | MikeJ[Laptop] | dialtone that is |
| 13:13.03 | ManxPower | key2: then use a voicemodem! |
| 13:13.06 | key2 | MikeJ[Laptop]: so If I put two FXO card on the same line and once I sent the flash, the second FXO card takes the new line, would that work ? |
| 13:13.09 | MikeJ[Laptop] | if you want to have an extension that automatically dials 2 specific numbers and 3ways them, throught the pots lines 3way, you can do that easy |
| 13:13.14 | ManxPower | Asterisk was never designed to do what you want to do. |
| 13:13.16 | MikeJ[Laptop] | no.. |
| 13:13.33 | bjohnson | key2: one fxo per line |
| 13:13.46 | MikeJ[Laptop] | maybe not easy.... |
| 13:14.00 | bjohnson | key2: it would be much easier to just use a voip provider for outgoing second line .. or get a second telco line |
| 13:14.03 | key2 | MikeJ[Laptop]: so how? |
| 13:14.14 | MikeJ[Laptop] | maybe not easy, but it is dooable |
| 13:14.33 | MikeJ[Laptop] | but it will not behave like your current 3way does fromthe user side |
| 13:14.50 | MikeJ[Laptop] | look on the wiki for x100p call waiting, you will see the hack. |
| 13:14.57 | ManxPower | This is so sad. Kansas Board of Education is going to have a vote today to decide if they are going to teach the Theory of Evolution in schools or not. |
| 13:15.19 | key2 | I basically don't pay my communication on this line here, and I wanted to know how I could set a callback to my cellphone so when I call asterisk, it calls me back, get the number I want to dial and call the otherone back in 3 way conferencing... |
| 13:15.32 | key2 | MikeJ[Laptop]: url ? |
| 13:15.37 | MikeJ[Laptop] | ManxPower, of course, there is always a good place for learning fiction in our schools |
| 13:15.39 | MikeJ[Laptop] | ~rtfw |
| 13:15.40 | jbot | hmm... rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 13:15.43 | Blackvel | so? Is the attended transfer issue now solved? :) |
| 13:15.59 | MikeJ[Laptop] | I think they call it english class |
| 13:18.04 | ManxPower | It's pretty obvious that key2 is not willing to read the Wiki |
| 13:18.24 | Blackvel | me neither |
| 13:18.27 | Blackvel | kick us both |
| 13:18.33 | Blackvel | :) |
| 13:18.40 | ManxPower | Don't tempt me. |
| 13:18.50 | Blackvel | ROFL |
| 13:20.03 | MikeJ[Laptop] | well.. their loss... |
| 13:21.03 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:21.34 | onkeltimm | Blackvel: I got me a mantis login... will look into it. |
| 13:21.58 | onkeltimm | if i can do the backport. |
| 13:22.19 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 13:22.30 | Blackvel | well, what shall I see. I wish you much fun the next weeks :) |
| 13:22.34 | Blackvel | I say. |
| 13:22.45 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 13:23.14 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 13:25.19 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 13:25.56 | kimc | any Cisco 7960 people around this morning ? |
| 13:25.57 | Zeeek | anyone hear of a problem where all of a sudden "reload" or "sip reload" take a very long time? |
| 13:26.08 | *** join/#asterisk iq (~iq@63-230-47-122.omah.qwest.net) |
| 13:26.14 | Zeeek | and initial startup as well? |
| 13:26.16 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:26.28 | ariel_ | Hello everyone |
| 13:26.47 | ManxPower | Zeeek: Yes. |
| 13:27.00 | Zeeek | where should I look for the wisdom? |
| 13:27.02 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 13:27.02 | ManxPower | stop using DNS names in Asterisk and use IP addresses. |
| 13:27.02 | kimc | Trying to get the voicemail button to work on a 7960 |
| 13:27.11 | Zeeek | ah wait a sec... |
| 13:27.19 | Zeeek | whoa |
| 13:27.33 | Zeeek | ManxPower you may be a genius |
| 13:27.50 | Zeeek | after DSL probs, maybe bad DNS... makes sense |
| 13:28.24 | ManxPower | Zeeek: Yes, I am. 8-) |
| 13:28.49 | *** join/#asterisk Senty (~senty@82.153.131.183) |
| 13:30.08 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 13:30.34 | Senty | can anyone help me with looping back between two e1's on a 410p? |
| 13:31.06 | Archi | morning. |
| 13:31.21 | Senty | afternoon :D |
| 13:31.45 | onkeltimm | yeah, definitely afternoon. don't be so us centric ;) |
| 13:32.02 | HeadachesAbound | just remember that somewhere in the world it is already tomorrow. |
| 13:32.04 | Archi | anybody have a link to good tutorial on interfacing with a PRI and attaining DID numbers? |
| 13:32.21 | Archi | Well I just got up, so for me it's 100% morning.. :) |
| 13:32.48 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:32.57 | Senty | can anyone help with looping back between two e1's on a 410p? |
| 13:33.22 | tzanger | ok it's my birthday, who's getting me what? |
| 13:33.35 | tzanger | and before anyone suggests it, I've already got enough swift kicks in the ass |
| 13:33.38 | Archi | :) |
| 13:33.54 | Archi | oh I was thinking spankings.. |
| 13:34.00 | tzanger | no bkw's got that covered |
| 13:34.16 | UltraGra | senty - yo! |
| 13:34.19 | Zeeek | anyone here have a teliax DID ? |
| 13:34.27 | Senty | hi gra |
| 13:35.03 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:35.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:35.06 | MikeJ[Laptop] | tzanger, err.. ummm |
| 13:35.26 | MikeJ[Laptop] | hey bkw_, tzanger says you have a lock on spanking him??? |
| 13:36.01 | tzanger | yeah it's my bday |
| 13:36.06 | Senty | anyone know how to get round the error == Everyone is busy/congested at this time |
| 13:36.36 | onkeltimm | tzanger: i'm getting you the unprecedented opportunity of backporting attended transfers to stable. sound good? |
| 13:36.52 | dmccollum | Sentry that means you don't have the zapta.conf setup correctly or the dial plan isn't setup right. |
| 13:36.56 | MikeJ[Laptop] | Senty, traffic cops! |
| 13:37.17 | Senty | dmccollum pvt? |
| 13:38.17 | tzanger | onkeltimm: sure; I don't use stable :-) |
| 13:38.24 | tzanger | Senty: yeah have more lines :-) |
| 13:38.33 | MikeJ[Laptop] | people keep saying backport, but no one is saying cash... hmmmmm |
| 13:38.43 | onkeltimm | Senty: hang up your phone, perhaps ;) |
| 13:39.16 | Archi | well at anyrate, happy birthday to you! |
| 13:39.26 | Senty | very good okeltimm, never thought of that ;) |
| 13:39.39 | *** join/#asterisk urmelZ (~Miranda@u16-13.dsl.vianetworks.de) |
| 13:40.29 | MikeJ[Laptop] | Senty, see, that is what we are here for |
| 13:41.53 | *** join/#asterisk sob0l (~sobol_@uo166.internetdsl.tpnet.pl) |
| 13:42.48 | tzanger | thanks Archi :-) |
| 13:43.03 | key2 | MikeJ[Laptop]: is asterisk usable with a normal modem ? or only zaptel ? |
| 13:43.11 | MikeJ[Laptop] | key2: |
| 13:43.14 | MikeJ[Laptop] | ~rtfw |
| 13:43.15 | jbot | rtfw is, like, Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 13:43.26 | MikeJ[Laptop] | ~docs |
| 13:43.27 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:43.34 | ManxPower | Well, I guess it's time for a /ignore again |
| 13:43.46 | MikeJ[Laptop] | the second is probably a little more civalized, isn't it... |
| 13:44.05 | *** join/#asterisk apardo (~apardo@80-28-161-183.adsl.nuria.telefonica-data.net) |
| 13:44.34 | Senty | anyone have any experience with digium te405/410 cards? |
| 13:44.52 | Archi | Is there there effort out there to allow people to become gateway's themselves and then benefit financially of it out there at the moment? If not, would that be something people would be interested in? |
| 13:45.58 | *** join/#asterisk durex (~ironman@200.101.109.138) |
| 13:46.00 | tzanger | jbot, wwjd is WWJD? JWRTFM!!! |
| 13:46.01 | jbot | ...but wwjd is already something else... |
| 13:46.11 | tzanger | jbot, forget wwjd |
| 13:46.11 | jbot | i forgot wwjd, tzanger |
| 13:46.13 | tzanger | jbot, wwjd is WWJD? JWRTFM!!! |
| 13:46.14 | jbot | okay, tzanger |
| 13:46.44 | Weezey | heh |
| 13:46.51 | MikeJ[Laptop] | that's wrong... |
| 13:46.59 | MikeJ[Laptop] | and that's from a jew. |
| 13:47.15 | Senty | has anyone done a loopback on e1's with asterisk for testing? |
| 13:47.15 | tzanger | you're Jewish? I have a question then |
| 13:47.19 | tzanger | I've never understood this |
| 13:47.32 | ManxPower | Anyone know what the Ultimate Jewish Dilemma is? |
| 13:47.34 | tzanger | if jesus was supposed to be the saviour and king of the jews, why do they hate him so much? |
| 13:47.43 | ManxPower | Free Bacon |
| 13:47.46 | Weezey | does jbot listen to everyone or only you tzanger? |
| 13:47.57 | tzanger | Weezey: find out yourself :-) |
| 13:47.59 | ManxPower | Weezey: jbot only listens to The Cool People |
| 13:48.06 | Weezey | jbot, make me coffee. |
| 13:48.09 | jbot | make: *** No rule to make target `me coffee.'. Stop. |
| 13:48.16 | Weezey | heh. |
| 13:48.47 | Weezey | jbot, vomit is available by calling 1-877-MY-VOMIT |
| 13:48.49 | jbot | ...but vomit is already something else... |
| 13:48.56 | MikeJ[Laptop] | christians beleive jesus is the savior... making them, well christians, jews don't by that whole jesus is the savior thing.... |
| 13:49.03 | *** part/#asterisk tabmoW (tabmow@tabmow.linuxfordummies) |
| 13:49.05 | durex | jbot, make me sleep |
| 13:49.07 | jbot | make: *** No rule to make target `me sleep'. Stop. |
| 13:49.07 | durex | ;-) |
| 13:49.20 | Weezey | jbot, make install |
| 13:49.22 | jbot | make: *** No rule to make target `install'. Stop. |
| 13:49.23 | MikeJ[Laptop] | but mostly it was the powermongers that didn't like jesus, not jews in general.. |
| 13:49.33 | tzanger | MikeJ[Laptop]: ok so why didn't hte christians say jesus was the saviour of the christians? |
| 13:49.41 | tzanger | it's weird |
| 13:49.44 | ManxPower | Weezey: There's nothing wrong with playing with the bot, as long as it's done in private and you wash your hands after. |
| 13:49.49 | MikeJ[Laptop] | don't ask me |
| 13:49.49 | Weezey | hah |
| 13:49.52 | onkeltimm | jbot, backport attended transfers for me, please |
| 13:50.17 | ManxPower | tzanger: I gave up trying to understand anything religios a long time ago |
| 13:50.19 | tzanger | jbot, eat onkeltimm |
| 13:50.21 | jbot | ACTION eats onkeltimm and falls over dead |
| 13:50.25 | onkeltimm | i KNEW he couldnt do that |
| 13:50.55 | tzanger | why are you heavy? |
| 13:51.02 | MikeJ[Laptop] | jbot, lart onkeltimm |
| 13:51.44 | onkeltimm | tzanger: no, german ;) |
| 13:51.44 | tzanger | where's that crazy sweede |
| 13:51.51 | tzanger | he can always make jbot violent |
| 13:52.06 | tzanger | oh wait he wasn't sweedish he was norwegian I think |
| 13:52.14 | tzanger | onkeltimm: your name doesn't give that away AT ALL |
| 13:52.26 | tzanger | the only way it could be more obvious was if it were onkelhans |
| 13:54.20 | Zeeek | ok, does anyone know about nufone codecs? |
| 13:54.32 | onkeltimm | or, like, onkültimm |
| 13:54.39 | onkeltimm | önkütümm |
| 13:54.54 | ManxPower | Zeeek: What about them? |
| 13:55.12 | Zeeek | heh, ManxPower you may be the savior again here |
| 13:55.12 | tzanger | only if I have that charset :-) |
| 13:55.18 | tzanger | önkütümm |
| 13:55.25 | Zeeek | all of a sudden it appears to only want to do g729 |
| 13:55.35 | Zeeek | is that something that may have changed recently? |
| 13:55.43 | ManxPower | Zeeek: What if you disallow=g729 in iax.conf? |
| 13:55.59 | ManxPower | Zeeek: Nufone totally redesigned their network to scale better. |
| 13:56.07 | Zeeek | it still wants to do g729 (which I have licensed but don't wanna use for these calls) |
| 13:56.16 | Archi | spaeking of codecs, has anyone implemented AT&T natural voices in Asterisk yet? |
| 13:56.21 | Zeeek | yeah and I fear there is a forced codec on their end now |
| 13:56.23 | tzanger | ManxPower: allow=all; disallow=g729 |
| 13:56.27 | fenlander | <!-- PayPal Logo --><table border="0" cellpadding="10" cellspacing="0" align="center"><tr><td align="center"></td></tr> |
| 13:56.29 | onkeltimm | tzanger: no unicode? |
| 13:56.33 | durex | Archi I heard something about it in a few weeks ago... |
| 13:56.41 | ManxPower | Zeeek: I can do G726 to NuFone |
| 13:56.48 | tzanger | onkeltimm: nope |
| 13:56.57 | bkw_ | <Zeeek>it still wants to do g729 (which I have licensed but don't wanna use for these calls) |
| 13:56.58 | Zeeek | Manx I'm talking about DID, I shouldhave said |
| 13:57.00 | tzanger | I'm a dim-witted north american; I see no need for anything more than 7-bit ASCII. :-0 |
| 13:57.02 | bkw_ | read what he said |
| 13:57.04 | tzanger | er :-) |
| 13:57.10 | onkeltimm | lol |
| 13:57.11 | Zeeek | I wasn't clear |
| 13:57.19 | bkw_ | Zeeek, yes you were |
| 13:57.34 | Zeeek | when you order DID with them, you can specify codecs |
| 13:57.41 | bkw_ | yes |
| 13:57.46 | bkw_ | you just disallow/allow what you want |
| 13:57.48 | Zeeek | It would appear that the specified codec has changed |
| 13:57.48 | bkw_ | in the order you want |
| 13:58.02 | bkw_ | go read the iax.conf.sample |
| 13:58.04 | Zeeek | because calls that worked before are now complaining about g729 |
| 13:58.06 | bkw_ | it has more info. |
| 13:58.07 | ManxPower | I just got an inbound call from Nufone using Speex |
| 13:58.12 | Zeeek | done all that bkw_ |
| 13:58.43 | ManxPower | Zeeek: I believe that Nufone MAY have added G729 to many of their servers |
| 13:58.55 | ManxPower | where they didn't have it on most of their servers before. |
| 13:59.05 | Zeeek | but it looks like they ONLY want to talk g729 |
| 13:59.22 | Zeeek | we have no config interface to change the DID codecs |
| 13:59.45 | JonR800 | I just accepted a call from nufone and I don't have G729 installed. |
| 13:59.57 | Zeeek | and I'm leaving town tomorrow and wanted to use the DID |
| 14:00.00 | Zeeek | JonR800 what codecs have you allowsed? |
| 14:00.03 | ManxPower | Zeeek: do an IAX2 debug to see what codecs Nufone claims to support when an incoming call happens |
| 14:00.13 | Zeeek | ManxPower ok |
| 14:00.32 | JonR800 | Zeeek: only ulaw |
| 14:01.22 | Zeeek | I thouhgt they didn't do ulaw? |
| 14:01.48 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 14:01.49 | jief- | hello |
| 14:03.18 | jief- | When we transfer a call from one extension to another, the first line is still active. We have to select it and hang it up. Is there a way for this to be done automatically? |
| 14:03.41 | langals | Hi there...if I am using meetme..and my clients are dialling in with GSM codec, then will meetme be transcoding from GSM to ULaw and back, or does it pass GSM straigtht through? |
| 14:03.45 | Zeeek | ManxPower nufone seems to complain about all codecs except 729 |
| 14:03.47 | *** join/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
| 14:04.02 | Zeeek | May 5 16:03:20 WARNING[1213]: channel.c:270 ast_best_codec: Don't know any of 0xf800 formats |
| 14:04.02 | Zeeek | May 5 16:03:20 ERROR[1213]: chan_iax2.c:5777 socket_read: No best format in 0xf800??? |
| 14:04.02 | Zeeek | May 5 16:03:20 NOTICE[1213]: chan_iax2.c:5779 socket_read: Rejected connect attempt from |
| 14:04.03 | bkw_ | langals, meetme speaks signed linear |
| 14:04.04 | JonR800 | what is format = 4? |
| 14:04.11 | bkw_ | ulaw |
| 14:04.30 | JonR800 | Zeeek: then yes they support ulaw :) cause that's what my call went through as. |
| 14:04.37 | JonR800 | thanks bkw |
| 14:04.42 | Zeeek | May 5 16:03:20 WARNING[1213]: channel.c:270 ast_best_codec: Don't know any of 0xf800 formatssomething is screwy here |
| 14:05.24 | Blackvel | ehm |
| 14:05.32 | Zeeek | if g729 is allowed, my asterisk complains about transcoding. If g729 is NOT allowed, nufone complains about not being codec compatible |
| 14:05.36 | Blackvel | agi_dnid = destination number |
| 14:05.50 | Blackvel | what is the agi variable for caller phone numberß |
| 14:05.53 | Blackvel | ? |
| 14:06.32 | Blackvel | zeeek: I believe something is weird in your system setup. I can not believe that nufone does not support ulaw/alaw |
| 14:06.57 | Blackvel | maybe nufone has some limitations for IAX connection? |
| 14:06.59 | Zeeek | no they do |
| 14:07.09 | langals | bkw - so it does do the transcoding then? |
| 14:07.13 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 14:07.49 | JonR800 | They do.. Zeeek, are you setting disallow / allow in general or under nufone? |
| 14:08.11 | Zeeek | under nufone - this config was working fine until a few days ago |
| 14:08.17 | JonR800 | ok |
| 14:08.44 | Zeeek | right now if I leave only ulaw I'm getting channel.c:270 ast_best_codec: Don't know any of 0xf800 formats |
| 14:09.02 | bkw_ | haha I know what your problem is |
| 14:09.04 | Zeeek | which makes me thing nufone is sending something funny? |
| 14:09.10 | Zeeek | TELL ME PLEASE!!!! :) |
| 14:09.12 | bkw_ | go rm -rf /usr/include/asterisk |
| 14:09.16 | bkw_ | and rebuild asterisk |
| 14:09.26 | bkw_ | you're doing stable? |
| 14:09.29 | Zeeek | yet I built this months ago? |
| 14:09.34 | Zeeek | ya 1.0.6 STABLE |
| 14:09.38 | bkw_ | ding ding ding |
| 14:09.42 | bkw_ | have you ever installed head? |
| 14:09.49 | Zeeek | long time ago |
| 14:09.55 | bkw_ | ding ding ding |
| 14:09.57 | bkw_ | rebuild |
| 14:09.57 | Zeeek | heh |
| 14:09.58 | bkw_ | go use cvs-hed |
| 14:10.01 | bkw_ | er head |
| 14:10.03 | bkw_ | it works |
| 14:10.04 | bkw_ | great |
| 14:10.09 | Zeeek | will do, will you explain while I do that? |
| 14:10.11 | JonR800 | im using stable and it works for me though |
| 14:10.45 | JonR800 | though im using stable from 3/2/05 |
| 14:11.34 | bkw_ | gcc -shared .libs/libchan_h323.la-2.o -L/usr/src/ooh323c_driver-0.1/ooh323c/src -lpthread -loostk -Wl,-soname -Wl,libchan_h323.so.1 -o .libs/libchan_h323.so.1.0.1 |
| 14:11.42 | tzanger | "ooh323" ? |
| 14:11.45 | bkw_ | it makes a call |
| 14:11.50 | bkw_ | but don't expect two |
| 14:12.04 | bkw_ | the stack capabilities are getting reset when the call gets destroyed |
| 14:12.08 | *** join/#asterisk Donuil (~fpatria@217.9.64.234) |
| 14:12.09 | Weezey | jbot, Three Point Four is a great song about finding a new vagina by The Bloodhoung Gang. |
| 14:12.10 | jbot | Weezey: okay |
| 14:12.15 | tzanger | hahahaha |
| 14:12.19 | tzanger | excellent |
| 14:12.34 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 14:12.36 | Zeeek | bkw_ do I need to make clean first? |
| 14:12.46 | Weezey | jbot, forget Three Point Four |
| 14:12.46 | jbot | Weezey: i forgot three point four |
| 14:12.51 | Weezey | jbot, Three Point One Four is a great song about finding a new vagina by The Bloodhoung Gang. |
| 14:12.53 | jbot | okay, Weezey |
| 14:13.00 | bkw_ | Zeeek, yes |
| 14:13.10 | bkw_ | jbot go to hell |
| 14:13.11 | jbot | [go to hell] a song |
| 14:13.15 | bkw_ | haha |
| 14:13.17 | bkw_ | stupid bot |
| 14:13.39 | Zeeek | bkw_ while I'm making, what made you see the problem? |
| 14:14.00 | bkw_ | Don't know any of 0xf800 formats |
| 14:14.01 | bkw_ | that |
| 14:14.05 | Zeeek | aha |
| 14:14.14 | Zeeek | I googled and only found like one reference |
| 14:14.16 | bkw_ | it might or might not be the issue.. but chances are |
| 14:14.24 | *** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net) |
| 14:14.27 | Zeeek | funny how it just started though |
| 14:14.34 | Donuil | hi to all... I have installed Zaptel Digium card correctly... I would to call my sip phone directly theough asterisk... I have associated an exten 5551 to my sip phone How can I do?I try exten => 1000,1,Goto(default,${EXTEN:1},1) |
| 14:14.50 | zoa | brian did you have any issues with the new sip jb ? |
| 14:14.56 | bkw_ | zoa not really |
| 14:15.03 | zoa | did you use grandstreams ? |
| 14:15.04 | bkw_ | Donuil, how long have you been using asterisk? |
| 14:15.07 | bkw_ | zoa now |
| 14:15.09 | bkw_ | zoa, cisco |
| 14:15.15 | zoa | we just had a reports of grandstream problems |
| 14:15.23 | zoa | although we tested a gs |
| 14:15.25 | bkw_ | exten => 5551,1,Dial(SIP/something) |
| 14:16.32 | MeTaBSD | Anyone use asterisk with Babytel here ? |
| 14:16.42 | Donuil | bkw_ is it important?^_^ however about a mounth |
| 14:16.45 | *** join/#asterisk convey (~kvirc@63.115.106.66) |
| 14:16.47 | MeTaBSD | and how can i flush i ghost channel CONGESTION ? |
| 14:16.59 | jief- | ok, if i get a call on my extention on line1. and i transfert it to another extension, its going to use line2 to do it. but its not hanging up line1 after the call is transfered. is there a way to do this automatically? is it me that's missing some config somewhere? |
| 14:17.21 | Zeeek | ManxPower thanks for your DNS hint. I changed providers and the DNS was only working on the fourth server. Naturally everything was sllllloooooowwwwww |
| 14:17.35 | Donuil | bkw. yes but I want to call from pstn.. |
| 14:17.53 | Weezey | That's a great point... If God didn't believe in himself would that be blasphemy? |
| 14:18.06 | bkw_ | Donuil, if you have used aserisk a month you should know this by now. |
| 14:18.22 | bkw_ | Donuil, what kind of card is hooked to the PSTN? |
| 14:18.48 | bkw_ | I suspect exten => s,1,Dial(SIP/something) |
| 14:18.50 | Donuil | tdm 400p... with two fxs channels and two fxo channels... |
| 14:18.50 | bkw_ | would work |
| 14:19.16 | bkw_ | the extensions.conf that comes with asterisk has all these examples in it |
| 14:19.22 | bkw_ | just takes some time to look at it an dlearn it |
| 14:19.53 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 14:20.16 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 14:20.19 | file[class] | hello folks |
| 14:20.53 | Zeeek | bkw_ that didn't fix it |
| 14:22.35 | Zeeek | Folks, setting only ulaw for nufone, outgoing works fine but I get this 0x800 codec message with incoming |
| 14:22.52 | zoa | 0x800ccc91 ? |
| 14:22.56 | zoa | :) |
| 14:23.06 | bkw_ | Zeeek, crank up debug |
| 14:23.09 | zoa | oh no outlook error messages in asterisk! |
| 14:23.21 | bkw_ | why do people run stable? |
| 14:23.25 | bkw_ | its not "stable" |
| 14:23.26 | Zeeek | iax2 debug? |
| 14:23.29 | bkw_ | it just feature freeze |
| 14:23.39 | ManxPower | bkw_: and bug fixes. |
| 14:23.41 | bkw_ | head surpasses stable in stability on alot of days |
| 14:23.45 | Zeeek | how can it not be stable when I haven't touched it in 2 months? :) |
| 14:23.59 | bkw_ | Zeeek, I suspect a sable -> head incompatibility |
| 14:24.12 | Zeeek | haven't used head in a long time |
| 14:24.12 | bkw_ | if Nufone is running head |
| 14:24.15 | bkw_ | it migh thave some issues |
| 14:24.15 | Blackvel | zeeek: do you do incoming or outgoing? [general] is always for incomnig (at least for sip.conf) |
| 14:24.16 | Zeeek | aha |
| 14:24.40 | ManxPower | Blackvel: BZZZT! Wrong! |
| 14:24.40 | bkw_ | Blackvel, that means nothing |
| 14:24.40 | Zeeek | ^^^^ bkw_ |
| 14:24.43 | _Thorr_ | hi, can i use an external dialup modem as a FXO port? |
| 14:24.46 | Blackvel | hm? |
| 14:24.51 | bkw_ | _Thorr_, NO |
| 14:24.55 | bkw_ | NEXT!!! |
| 14:24.58 | Zeeek | All I can say is, other DID are working in iax |
| 14:25.00 | Blackvel | but it works in that way for me..sorry |
| 14:25.09 | Zeeek | so WHY has nufone suddenly changed? |
| 14:25.14 | bkw_ | Zeeek, if jerjer updated their might very well be some incompatibility |
| 14:25.20 | zoa | nufone changes every day probably |
| 14:25.24 | ManxPower | Zeeek: Try this. Put context=INVALID in [general] then put the correct context= line in each of the iax.conf entries. If incoming calls are sent to INVALID, then the incoming call is not authenticating correctly. |
| 14:25.25 | Zeeek | the default stuff is only by ummmm default |
| 14:25.41 | Zeeek | k |
| 14:25.45 | _Thorr_ | bkw_, can you explain? |
| 14:25.48 | jakepdev | anyone know how to get a peak zap channel usage report |
| 14:25.51 | bkw_ | its saying unknow format |
| 14:26.00 | bkw_ | OK PEOPLE |
| 14:26.05 | bkw_ | _Thorr_, search google. |
| 14:26.08 | newl | Since when does a 'no' answer require explaination? :) |
| 14:26.09 | bkw_ | jakepdev, search google. |
| 14:26.17 | file[class] | the teacher is sitting right behind me |
| 14:26.25 | _Thorr_ | bkw_, ok, thx |
| 14:26.30 | bkw_ | file[class], you're teacher is stupid! |
| 14:26.31 | bkw_ | haha |
| 14:26.34 | bkw_ | er your |
| 14:26.36 | bkw_ | fuck I can't type |
| 14:26.38 | file[class] | I just minimized my window so that he can't see |
| 14:26.43 | file[class] | thank god this is encrypted |
| 14:26.50 | bkw_ | ROT26 |
| 14:27.22 | Blackvel | file[class]: what is encrypted? |
| 14:27.29 | Blackvel | what class are you taking? |
| 14:27.31 | Zeeek | ~seen JerJer |
| 14:27.38 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 7d 9h 3m 28s ago, saying: 'try ulaw or gsm'. |
| 14:27.43 | *** join/#asterisk moy (~kvirc@dsl-200-95-105-36.prod-infinitum.com.mx) |
| 14:27.44 | newl | Does that camera up in the corner recording you know that your communication is encrypted? :D |
| 14:28.11 | bkw_ | its watching the keyboard via a keylogger |
| 14:28.16 | JonR800 | must be HS if it's that important to keep it secret. |
| 14:28.20 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 14:28.20 | newl | bing! |
| 14:28.41 | jief- | i have to specifie t,T as options for an extension for the caller/called to be able to transfer a call to someone else right? |
| 14:28.45 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 14:28.59 | jief- | like this exten => 111,1,SIP/501(Dial,10,t,T) ? |
| 14:29.02 | Bile_One | Good Morning everyone. |
| 14:29.04 | file[class] | this is HS |
| 14:29.04 | ManxPower | jief-: only if the phone you are using is too stupid to have it's own transfer feature |
| 14:29.23 | JonR800 | heh |
| 14:29.24 | Bile_One | Morning ManxPower. |
| 14:29.38 | Zeeek | this sucks, the day before I leave |
| 14:29.39 | jief- | ManxPower: it has a transfer feature, but when i transfer it, it won't hangup the first line that it received the call on |
| 14:29.46 | ManxPower | jief-: There are so many things wrong with your Dial line I can't even begin to start to point them out. |
| 14:29.54 | JonR800 | I remember one time, getting kicked out of the labs because I ssh'd into home.. because the teacher didn't know what i was doing.. lameness |
| 14:30.08 | jief- | ManxPower: enlighten me |
| 14:30.13 | ManxPower | "This one time as Band Camp...." |
| 14:30.19 | JonR800 | The one time I'm actually trying to use the computers to turn in homework. |
| 14:30.26 | ManxPower | jief-: Dial(SIP/501,10,tT) |
| 14:30.35 | jief- | ManxPower: i know, that was a mistake |
| 14:30.39 | jief- | in extensions.conf its ok |
| 14:30.44 | newl | ManxPower: Please step away from the pie and keep your hands where we can see them. |
| 14:30.45 | jief- | thats what i meant |
| 14:30.49 | ManxPower | jief-: your mistakes on this channel waste people's time. |
| 14:31.02 | Zeeek | http://pastebin.ca/11033 |
| 14:31.24 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:31.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.26 | jakepdev | bkw - i checked google - is it an asterisk command - or a seperate app? |
| 14:31.30 | Blackvel | file[class]: HS? |
| 14:31.34 | jief- | ManxPower: you are such a nice person |
| 14:31.41 | Blackvel | hi jakepdev |
| 14:31.48 | bkw_ | jakepdev, you'll need to setup mrtg or something to monitor channel usage |
| 14:31.49 | jakepdev | hey Blackvel |
| 14:31.51 | bkw_ | MOVE ALONG NOW |
| 14:31.56 | jakepdev | thanks d00d |
| 14:31.59 | bkw_ | ;) |
| 14:32.16 | Blackvel | do you US guys work today? |
| 14:32.33 | jakepdev | yep |
| 14:32.34 | ManxPower | Blackvel: No. We take the day off when the UK has elections. |
| 14:32.35 | convey | What is the VoIP carrier of chice for the US? |
| 14:32.40 | newl | can't see why they wouldn't. :) |
| 14:33.07 | Blackvel | germans dont |
| 14:33.12 | Blackvel | except me |
| 14:33.17 | Bile_One | convey none |
| 14:33.31 | jakepdev | bkw - just to confirm - mrtg works for zap channel usage? |
| 14:33.43 | newl | ManxPower: You'd need to take the day off to clean up the UK embassey. :) |
| 14:33.48 | file[class] | I was found out! |
| 14:33.51 | convey | So they all basically have lousy service? |
| 14:33.52 | file[class] | my teacher saw my Skype window |
| 14:33.55 | ManxPower | Ah, today, 60 years ago, what the liberation of Aushwitz |
| 14:34.15 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 14:34.23 | Zeeek | ok, wait a sec: requested/capability 0x100/0xf900 incompatible with our capability 0xf804 |
| 14:34.39 | JonR800 | jakepdev: http://www.voip-info.org/tiki-index.php?page=Asterisk%20monitoring |
| 14:34.41 | Blackvel | file[class]: skype? what are you talking about? :) you use skype instead of irc? |
| 14:34.50 | Zeeek | 4 is ulaw, 800 ??? |
| 14:34.51 | Blackvel | you are using... |
| 14:34.58 | ManxPower | Zeeek: in [general] allow=all then in each section disallow=all and allow= the one codec you want to use. |
| 14:35.03 | Bile_One | convey, they all have issues. Bigest provider more than likely is Vonage. But that is open to discussion. |
| 14:35.20 | Blackvel | ManxPower: so [general] is not for incoming? |
| 14:35.26 | ManxPower | The most reliable VoIP carrier is your local telco |
| 14:35.31 | Blackvel | haha |
| 14:35.41 | ManxPower | Blackvel: [general] is for general and for UNAUTHENTICATED calls. |
| 14:35.53 | convey | Does Vonage work with *? |
| 14:36.10 | Zeeek | ManxPower no change after reload |
| 14:36.13 | Nivex | Vonage doesn't work with anyone. |
| 14:36.17 | Nivex | except themselves. |
| 14:36.36 | Blackvel | ManxPower: hm. seems I have insecure=very for all silly sip providers like sipgate, nikotel, fwd anyways |
| 14:36.38 | Zeeek | this has gotta be something nufine is "sending" |
| 14:36.42 | ManxPower | Zeeek: try a restart, I've seen sip.conf options not be updated on a reload |
| 14:36.45 | Nivex | someone should put that in the topic... it would save so much time. |
| 14:36.46 | Blackvel | not sure if calls are authenticated or not |
| 14:36.46 | Zeeek | true |
| 14:36.53 | ManxPower | or just unload chan_sip.so and load chan_sip.so |
| 14:37.02 | Zeeek | sip? why sip, it's iax |
| 14:37.09 | ManxPower | ..er...sorry |
| 14:37.13 | Blackvel | yeah sure |
| 14:37.17 | Blackvel | i just mean to ManxPower |
| 14:37.18 | Zeeek | and I usually do a restart in desperation |
| 14:37.19 | ManxPower | unload chan_iax2.so and load chan_iax2.so |
| 14:37.26 | Blackvel | because he corrected me in total |
| 14:37.41 | ManxPower | Blackvel: if you are accepting unauthenticated calls and don't know it, you have major problems |
| 14:37.56 | convey | So if all VoIP carriers are sub par, which is the best among them that is compatable with *? |
| 14:38.01 | Blackvel | hm |
| 14:38.03 | Blackvel | not sure |
| 14:38.05 | ManxPower | that's why I ALWAYS have context=INVALID in [general] of sip.conf and iax.conf |
| 14:38.08 | Zeeek | ManxPower well same ol : requested/capability 0x100/0xf900 incompatible |
| 14:38.16 | ManxPower | Blackvel: it was not a question |
| 14:38.25 | ManxPower | Zeeek: what codec are you allow= ing? |
| 14:38.27 | Zeeek | why would they berequesting 0x900? |
| 14:38.37 | Blackvel | ManxPower: happens that most providers only work with insecure=very |
| 14:38.46 | Zeeek | I've tried several combos right now only ulaw |
| 14:38.53 | Blackvel | back to coding.. |
| 14:39.02 | ManxPower | Blackvel: no, bad providers only work with insecure=very |
| 14:39.36 | jakepdev | JonR800 - much appriciated |
| 14:40.00 | file[class] | and away I go |
| 14:40.36 | Bile_One | Shoot callforwarding is driving me mad! |
| 14:40.38 | JonR800 | np |
| 14:42.43 | Bile_One | Hey ManxPower, you work with Andy Powell on app_cepstral.c? |
| 14:42.53 | ManxPower | Bile_One: no |
| 14:43.07 | Bile_One | ManxPower, I wish you would have.... |
| 14:43.20 | Zeeek | If I allow g729, asterisk will accept the call but: May 5 16:42:35 WARNING[8003]: channel.c:1879 ast_request: No translator path exists for channel type ZAP (native 68) to 256 |
| 14:43.27 | ManxPower | Bile_One: not with my programming skills, you don't |
| 14:43.34 | Bile_One | Andy's is very good. |
| 14:43.45 | ManxPower | Zeeek: don't accept G729 for that peer/user unless you want to use it. |
| 14:43.46 | jsharp | Zeeek: Do you have G729 licenses? |
| 14:43.48 | Zeeek | I think the issue seems to be forcing 729 |
| 14:43.53 | Zeeek | yes I have 4 lic |
| 14:44.15 | jsharp | And g729 shows up in "show translations" with a valid cost? |
| 14:44.17 | Zeeek | ManxPower the point is, if I accept it almost works, if not, it throws a 0x900 format |
| 14:44.33 | Zeeek | jsharp no no g729 trans |
| 14:45.06 | Zeeek | we have had a couple of power outages - could that have screwed the licenses? |
| 14:45.31 | Zeeek | should the translations costs be showing? (you imply yes) |
| 14:45.36 | ManxPower | Zeeek: if G729 doesn't show up with a cost in "show translations" then you don't have G729 license installed |
| 14:45.48 | Zeeek | well, they were |
| 14:45.54 | Zeeek | I think there is a way to reinstall |
| 14:46.08 | Bile_One | bkw_ will app_machinedetect.c be included inthe next release? |
| 14:50.06 | Zeeek | When does the "host id" change for these licenses? |
| 14:50.14 | Zeeek | ah I see.... |
| 14:50.24 | Zeeek | I changed the eth card a while back |
| 14:50.34 | *** join/#asterisk cmk (~cmk_@p54A3F09D.dip.t-dialin.net) |
| 14:51.07 | *** join/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 14:51.20 | queuetue | Hi, all. Do all of the voip providers have lots of downtime (4 or 5 times a day) or does Broadvoice just suck ass? |
| 14:51.30 | ManxPower | queuetue: Yes. |
| 14:51.50 | queuetue | ManxPower, yes, to all, or Yes, BV sucks ? |
| 14:51.56 | ManxPower | I've not had any problems with NuFone or Teliax, but I don't to a lot of calling using them. |
| 14:52.04 | ManxPower | queuetue: yes to both 8-) |
| 14:52.25 | Zeeek | Thanks guys, it was a long tortured path but there it is. I changed a card, that voided the g729 registration and the rest is history |
| 14:52.39 | Bile_One | ManxPower, any idea why I can't get a SIP extenstion to Call forward to a ZAP extension and have it work through call files? |
| 14:52.56 | ManxPower | Bile_One: because your SIP phone is eating the *XX codex |
| 14:52.58 | ManxPower | Bile_One: because your SIP phone is eating the *XX codes |
| 14:53.22 | Bile_One | *XX codes? |
| 14:53.41 | queuetue | ManxPower, so, VOIP is unreliable in general? |
| 14:53.51 | queuetue | (using commercial carriers) |
| 14:53.52 | Bile_One | Not VOIP just the providers |
| 14:54.05 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 14:54.06 | *** join/#asterisk mansol (~blah@210-9-153-152.msi.net.au) |
| 14:54.26 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:54.27 | queuetue | So how do you make it work? |
| 14:54.43 | queuetue | Phones are pretty mission-critical... |
| 14:55.16 | Bile_One | Since VoIP is so dependent on the network it is as good as it gets for now. Mark Spencer said that Cell Phones were great for VoIP because they lower everyone's expectations. |
| 14:55.18 | mansol | hi everyone |
| 14:55.44 | ManxPower | queuetue: If you are relying on the internet of course it's not going to be reliable. |
| 14:55.58 | queuetue | ... What do you run voip over, then? |
| 14:56.05 | Bile_One | My Quote is from Marks' Keynote in Atlanta |
| 14:56.05 | ManxPower | queuetue: I run VoIP on my local LAN and my WAN, but to talk to the PSTN I use PRIs |
| 14:56.18 | newl | PtP PRI? :) |
| 14:56.18 | ManxPower | I also have a couple of VoIP providers that I use for overflow. |
| 14:56.43 | queuetue | Hrm. That sounds considerably less appealing, to us anyway. |
| 14:57.03 | ManxPower | Things would be somewhat more reliable if the VoIP providers themselves were more reliable, but that's only part of the issue. |
| 14:57.55 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 14:58.00 | Bile_One | queuetue, listen to ManxPower he knows what he is saying, backups, and alternate routing can save your life. |
| 14:58.24 | ManxPower | queuetue: You realize that Asterisk originally didn't even support VoIP, right? |
| 14:58.26 | JonR800 | or at least make it a lot easier when things do go wrong. |
| 14:58.30 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 14:58.38 | ManxPower | It originally supported the Tormeta cards for T-1/E-1 |
| 14:58.50 | Moonwick | you know, back in the stone age |
| 14:58.53 | Moonwick | what was that, 2004? :) |
| 14:59.01 | Bile_One | haa haa haa |
| 14:59.04 | onkeltimm | as for the attended transfer backport: far from trivial... makes use of loads of stuff that changed since stable... |
| 14:59.07 | queuetue | I suppose - It's unfortunate, though, because I'm looking for an inexpensive phone solution, not so much just a cheap PBX. |
| 14:59.15 | jsharp | Then Mark developed the ultimate voip protocol that doesn't suck. IAX. |
| 14:59.27 | ManxPower | jsharp: IAX isn't prefect. 8-) |
| 14:59.33 | Bile_One | but it isn't so good going out a zap device. |
| 14:59.37 | JonR800 | haha there's some hype. |
| 14:59.39 | ManxPower | I think IAX3 will be perfect. 8-) |
| 15:00.00 | Bile_One | And when 4 comes out, no one will use anything else. |
| 15:00.07 | jsharp | Not perfect, but mark definitely had his collective shit together when he wrote it. |
| 15:00.10 | Bile_One | ;) |
| 15:00.20 | ManxPower | It can address the lingering NAT issues with IAX and the call records with IAX native transfer. |
| 15:00.53 | ManxPower | jsharp: Yes, Mark is very smart, but I think luck played a part too. |
| 15:01.08 | jsharp | I think someone once said "h.323 was designed by telephony people with no concept of the internet...and SIP was designed by internet people with no concept of telephony" |
| 15:01.34 | ManxPower | jsharp: and RTP was designed by people that really should be publically shot. |
| 15:01.39 | ManxPower | But I don't feel strongly about the issue. |
| 15:01.39 | jsharp | And one of these days I'll actually get to meet Mark or at least hear him speak. |
| 15:02.03 | queuetue | Is IAX more momentary-dropout-resistant? |
| 15:02.03 | newl | I'd hate to see your strong feelings then. :) |
| 15:02.13 | Bile_One | haa haa haa |
| 15:02.16 | kimc | Anyone know how I can simulate an incoming call ? |
| 15:02.18 | tzanger | jsharp: he's cool |
| 15:02.23 | tzanger | and he likes his steaks rare |
| 15:02.53 | Bile_One | kimc yes add a context with an extension number |
| 15:03.00 | jsharp | Astricon US better be in Atlanta again this year, that way I can drive across town instead of cross country. |
| 15:03.22 | tzanger | newl: like mark then |
| 15:03.25 | tzanger | I like mine medium rare |
| 15:03.28 | tzanger | little bloody |
| 15:03.33 | kimc | Does the context need to exist anywhere else ? |
| 15:04.10 | kimc | Anywhere in extensions.conf ? |
| 15:04.18 | Bile_One | kimc in extensions.conf |
| 15:04.27 | kimc | Ok thanks muchly |
| 15:04.28 | Bile_One | then s,1 blah |
| 15:04.37 | kimc | right |
| 15:04.50 | Bile_One | well really s,1,blah |
| 15:05.10 | ManxPower | It's looking like my PSTN provider lets me set callerid name. |
| 15:05.29 | Bile_One | ManxPower, can you be so kind as to explain why SIP is eating my *XX's? |
| 15:05.29 | ManxPower | Of course it only works within their network. |
| 15:05.33 | tzanger | ManxPower: bell canada does that |
| 15:05.38 | cjk_ | ok txfax is working to some fax machines, but not to mine...... anyone have seen similar problems? |
| 15:05.48 | ManxPower | Bile_One: I can't. Check the instrucitons for your phone. |
| 15:05.58 | newl | Bile_One: Because it's rated NC-17? |
| 15:06.00 | ManxPower | cjk_: you have an old spandsp? |
| 15:06.04 | newl | quack! |
| 15:06.38 | jsharp | Speaking of fax? is anyone taking up that tasty bounty for T.38 in *? |
| 15:06.39 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:06.39 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:06.55 | eKo1 | How tasty? |
| 15:07.32 | jsharp | $4500 |
| 15:07.38 | ManxPower | Isn't the T.38 bounty up to like US$5,000? |
| 15:07.49 | jsharp | Hence the "tasty" description. |
| 15:08.21 | newl | I'd have to wait for the AUD to drop to $.55 for something like that to be worthwhile hehe |
| 15:08.27 | ManxPower | $4500 doesn't sound all that tasty to me. Not considering it's for T.38 |
| 15:08.49 | Bile_One | ManxPower, so if I set callforwarding up with Xten, it stays registered to that IP, and really never calls the ZAP phone? |
| 15:08.50 | eKo1 | Man, why are people bothering with faxing. Haven't they heard of e-mail. |
| 15:08.57 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 15:09.09 | jief- | eKo1: agreed |
| 15:09.13 | ManxPower | eKo1: signed contracts is what we use it for. |
| 15:09.19 | jsharp | Cause email still isn't universally accepted as a signed document. |
| 15:09.23 | ManxPower | thousands and thousands of signed contracts per year. |
| 15:09.41 | newl | jsharp: That depends on the country in which you live and its laws. |
| 15:09.41 | eKo1 | Just digitally sign the e-mail docs and it's done. |
| 15:09.48 | JonR800 | plus it's easier to send paper through fax than through email :) |
| 15:09.59 | jsharp | That's why I said "universally" |
| 15:10.02 | ManxPower | Plus the fact that at lest %50 of my users don't even have a computer at the office. |
| 15:10.05 | Bile_One | eKo1, then you have to have a CA and it gets nasty who owns the control |
| 15:10.08 | JonR800 | if you have a document on hand do you really want to scan it in and email it? |
| 15:10.18 | newl | jsharp: :) |
| 15:10.33 | eKo1 | Haven't people heard of .doc or .pdf? |
| 15:10.40 | eKo1 | c'mon |
| 15:10.48 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 15:10.49 | jsharp | There's still a bunch of people on Omicron Persei 8 that won't accept emails. |
| 15:11.05 | JonR800 | not when they have the sheet of paper in hand ready to be signed. |
| 15:11.12 | newl | And a few more on smoke signals.. |
| 15:11.15 | *** join/#asterisk apardo (~apardo@80-28-161-228.adsl.nuria.telefonica-data.net) |
| 15:11.38 | eKo1 | JonR800: well, then they need to stop wasting paper. |
| 15:11.46 | Bile_One | eKo1, but I can alter a .doc, and a .pdf |
| 15:12.03 | jsharp | You can alter a faxed document too. |
| 15:12.08 | JonR800 | lol.. true, wasn't the paperless office supposed to happen 10-20 years ago? |
| 15:12.11 | eKo1 | You can alter any kind of document. |
| 15:12.14 | tzanger | Bile_One: so sign it |
| 15:12.19 | tzanger | it's not fucking hard |
| 15:12.21 | onkeltimm | huh? whats that i can talk to another phone normally but i cannot hear the demo sounds played? |
| 15:12.25 | Bile_One | jsharp, yes but I have an original right? |
| 15:12.43 | ManxPower | I'll be on #asterisk stable if anyone needs help |
| 15:12.46 | ManxPower | ..er. |
| 15:12.49 | ManxPower | I'll be on #asterisk-stable if anyone needs help |
| 15:12.51 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 15:12.53 | Bile_One | ManxPower, I need it |
| 15:13.08 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 15:13.37 | *** join/#asterisk SirPrize (~blah@host-84-9-107-20.bulldogdsl.com) |
| 15:14.59 | eKo1 | It seems NewSole emphasized coffee and smokes for some reason. |
| 15:15.01 | onkeltimm | echotest also not working what did i do wrong now? |
| 15:15.12 | Juggie | pastebin your extensions newsole |
| 15:16.10 | gaba | hi |
| 15:16.41 | SirPrize | Could someone please tell me what I should configure so I can place a call through my * server, to SipGate ? |
| 15:17.25 | NewSole | well I have to step out for coffee I just woke up.... get brain moving |
| 15:17.52 | *** join/#asterisk lespiggot (~lespiggot@zeus.greavessports.com) |
| 15:18.01 | SirPrize | I already have registered for incoming SIP calls, and they work fine. I can also make calls to other people who are also on my provider (GossipTel) |
| 15:18.34 | SirPrize | but how should things be configured so that I can make a call to SipGate via my Asterisk server? Surely this doesn't need to go through GossipTel ? |
| 15:18.35 | sudhir492 | With cvs head, music on hold seems to be broken (or I have the incorrect version of mgp123) |
| 15:18.44 | bkw_ | works h ere |
| 15:19.08 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 15:19.27 | sudhir492 | bkw_: what version of mpg123 do you have |
| 15:19.54 | onkeltimm | i dial into *, * says it is playing demo sounds but i cannot hear them... whats wrong? normal calls work (1.0.7) |
| 15:19.55 | bkw_ | 0.59r |
| 15:20.14 | sudhir492 | let me check mine |
| 15:20.23 | cjk_ | mansol: |
| 15:20.47 | sudhir492 | mine is mpg123-pre0.59s |
| 15:21.26 | Gand_DJ | Virbiage (firefly) has released their customized version of the libIAX2 library |
| 15:21.32 | sudhir492 | that could be the problem |
| 15:22.07 | *** part/#asterisk lespiggot (~lespiggot@zeus.greavessports.com) |
| 15:22.52 | mansol | I am after some help. We are using a asterisk with a TDM04B to connect to a Hybrex phone system and act as a voicemail system. I can see the CallerID coming in but I am actually after the Called number! |
| 15:23.03 | Lee__ | are there any Debian voip-pkg members here? |
| 15:23.36 | mansol | is anyone here familiar with the internals of how the callerid works on the zap channel? |
| 15:25.01 | *** join/#asterisk packetman (~324@d141-12-203.home.cgocable.net) |
| 15:25.41 | tzanger | mansol: what do you mean |
| 15:26.03 | packetman | Anyone know a simple what of stoping the ZAP/1 Channel from picking up incoming PSTN calls. Im using Asterisk@home |
| 15:26.18 | SirPrize | How can I let my Asterisk make a call to any SIP address someone types in via their X-Lite phone ? |
| 15:26.22 | tzanger | packetman: assign it a context that has no answer |
| 15:26.38 | packetman | k |
| 15:26.40 | packetman | umm |
| 15:26.48 | packetman | little bit of a newbie |
| 15:27.19 | mansol | well, the callerid is being passed through from the phone system in the format: C11+STB+STA where STA is the calling extension and STB is the called number (the extension that was actually dialled) |
| 15:27.30 | packetman | so I would write a context with the no answer option and assign it to???? |
| 15:27.42 | SirPrize | that is, user 2000 on my asterisk server wants to call SIP users at other domains - what should I do to enable this ? |
| 15:27.59 | mansol | the problem is that asterisk correctly sees the STA portion and sets the CallerID to that value. However, I want the STB value.... |
| 15:28.04 | tzanger | packetman: you'd make a new context, say [noanswer] and make s,1,NoOp(ha ha can't pick up, pthbtbtbtbtbt!) |
| 15:28.11 | tzanger | and then say zap channel 1's context is [noanswer] |
| 15:28.18 | newl | SirPrize: place that user in a context that permits the calls. |
| 15:28.37 | packetman | sweet that make sence, thanks Tzanger |
| 15:29.06 | mansol | any ideas? |
| 15:29.07 | SirPrize | newl: I have a 'outgoingcalls' context, which uses GossipTel as my provider, but how do I make calls for example to SipGate? Surely this doesn't have to go through GossipTel ? |
| 15:29.55 | *** join/#asterisk RazorJack (~RazorJack@205.211.153.199) |
| 15:30.11 | newl | SirPrize: It'll go through whoever you tell it to go through first. |
| 15:30.19 | RazorJack | I have changed my rxgain/txgain in zapata.conf, relodaed asterisk and it does not seem to have any effect.... what else do I need todo? |
| 15:30.41 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfi7o.dialup.mindspring.com) |
| 15:31.00 | newl | SirPrize: or alternatively, you could use a global variable to determine the next provider to use and cycle between them, or enable some time based least call routing type of dial plan. |
| 15:31.01 | jsharp | Did you restart asterisk or just reload it? |
| 15:31.07 | SirPrize | can't my Asterisk server directly communicate with the MIT or SipGate or FWD server of the user that I am trying to call, without redirecting all calls through GossipTEl ? |
| 15:31.12 | RazorJack | jsharp: shut it down, load it up |
| 15:31.32 | SirPrize | newl: can't my Asterisk server directly communicate with the MIT or SipGate or FWD server of the user that I am trying to call, without redirecting all calls through GossipTEl ? |
| 15:32.20 | newl | SirPrize: sure thing. |
| 15:32.49 | SirPrize | newl: How would I do that? Because at the moment, all my outgoing calls go through GossipTel :( |
| 15:32.58 | *** join/#asterisk rkioko (~kiokorobe@196.200.26.42) |
| 15:32.58 | newl | SirPrize: My chosen config uses different extension numbers for different providers. |
| 15:33.26 | SirPrize | newl: different EXTENSION numbers ??? |
| 15:34.23 | HeadachesAbound | Anybody here ever used Festival Lite with *? |
| 15:34.41 | newl | SirPrize: yes. e.g. If I dial 9, my call is placed through a mates asterisk box, if I dial 8, I get FWD. 7 gets me firefly, and so on. |
| 15:34.41 | *** join/#asterisk packetman (~324@d141-12-203.home.cgocable.net) |
| 15:35.22 | SirPrize | newl: Ah, I see - using a different prefix for each provider. But what if in my x-lite softphone, I just enter an address: sip:username@somedomain.com |
| 15:35.57 | packetman | <tzanger> So I assume I would add this [noanswer] context in the extensions_custom.conf because I am using AMP? |
| 15:36.22 | newl | SirPrize: hmm..I don't have the answer for that one, sorry. I've never toyed with that scenario. |
| 15:36.33 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 15:37.16 | SirPrize | newl: thanks anyway. any pointers on what you think I could look at to find an answer to this ? |
| 15:37.19 | onkeltimm | could please anyone help me get my sounds back? i found nothing on the ml... |
| 15:38.16 | newl | SirPrize: try seaching the wiki, perhaps there may be an existing example. |
| 15:38.37 | SirPrize | Does anyone have an idea how I can configure my Asterisk to accept calls to addresses of the form sip:user@somedomain.com , directly contacting the somedomain.com SIP server ? SRV Lookups are already enabled by default, and haven't been disabled |
| 15:38.38 | newl | SirPrize: or google the mailing list archives. |
| 15:39.09 | SirPrize | newl: have already read through the Wiki pages, but haven't found one. Will check the mailing list next, thanks |
| 15:41.58 | packetman | Hi guys [custom-noanswer] |
| 15:41.58 | packetman | exten => s,1,NoOp(noanswer) |
| 15:42.09 | packetman | If I created a context like this [custom-noanswer] |
| 15:42.09 | packetman | exten => s,1,NoOp(noanswer) |
| 15:42.28 | packetman | and I wanted it to cause the Zap/1 channel not to answer where would I put it |
| 15:42.35 | tzanger | packetman: as I said |
| 15:42.46 | tzanger | tell asterisk that the context for the zap channel is [custom-noanswer] |
| 15:42.46 | onkeltimm | i rebooted the box and now PlayBack() is working! this scares me... |
| 15:42.55 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 15:43.11 | Blackvel | what is the asterisk caller variable in AGI? agi_dnid is the caller which is dailed, not the dailer number |
| 15:43.14 | eKo1 | Vodoo magic. |
| 15:43.34 | packetman | <Tzanger> Thanks I'm a little new, where would I tell asterisk, in what conf file |
| 15:43.48 | *** part/#asterisk SirPrize (~blah@host-84-9-107-20.bulldogdsl.com) |
| 15:43.53 | *** join/#asterisk file[physics] (~jcolp@66.199.241.90) |
| 15:43.57 | file[physics] | yay physics class |
| 15:43.59 | jakepdev | blackvel - agi_callerid |
| 15:44.03 | packetman | I've already created [the noanswer] |
| 15:44.35 | Atacomm | file: lol, you need a life outside of voip, go find some cute girls at school........LOL |
| 15:44.48 | packetman | would I place it in the zapata-auto.conf |
| 15:44.49 | Silik0n | or just surf some pr0n |
| 15:45.31 | jakepdev | packetman - zapata.conf is the default location |
| 15:45.42 | *** join/#asterisk Moc (~mochouina@h66-201-214-109.gtconnect.net) |
| 15:46.01 | jakepdev | packetman - but yours may be in zapata-auto.conf |
| 15:46.02 | file[physics] | Atacomm: they're right behind me |
| 15:46.14 | Atacomm | file: yeah, laughing |
| 15:46.24 | Atacomm | lol |
| 15:46.29 | file[physics] | yes, at you :P |
| 15:47.01 | Moc | hey Atacomm long time no see |
| 15:47.02 | Atacomm | they are like "josh, what a complete geek....lets go find someone strong and handsome"....hmm, like me :) LOL, okay maybe not |
| 15:47.09 | Atacomm | Hey Moc, how goes the battle? |
| 15:47.10 | file[physics] | I'm glad I actually have a half decent computer right here |
| 15:47.23 | packetman | thanks <jakepdev> I think I've found it |
| 15:47.31 | Moc | Atacomm: Which one ? |
| 15:47.31 | jakepdev | np |
| 15:47.37 | hardwire | hmmm |
| 15:48.06 | hardwire | 233mhz geode doing voice prompts is quite choppy.. I moved all the gsm files into a ramdisk.. still choppy.. |
| 15:48.22 | hardwire | the actual echo test as well as phone calls are working very well.. no chop.. |
| 15:48.32 | Qwell | transcoding the files? |
| 15:48.39 | hardwire | this is using gsm or g.729 for the call codec |
| 15:48.55 | hardwire | Qwell: well.. I would have thought gsm to gsm would work just fine |
| 15:49.08 | hardwire | but when I call in to test w/ the gsm codec.. its just as choppy |
| 15:49.18 | hardwire | so yeh.. I think its recoding all the prompts |
| 15:49.33 | hardwire | I wonder if I need to transcode them all as ulaw just tomake things easier |
| 15:51.12 | jakepdev | hardwire - i had issues with the choppy voice prompts when I had zap enabled but no zaptel cards installed. don't know if that's what you're seeing |
| 15:51.24 | hardwire | yes |
| 15:51.30 | hardwire | I need to use ztdummy for the conferencing |
| 15:51.45 | hardwire | let me get rid of chan_zap and see |
| 15:52.06 | jakepdev | i did a modprobe -r zaptel |
| 15:52.40 | *** join/#asterisk adjacent (~scott@nc-65-40-81-64.sta.sprint-hsd.net) |
| 15:52.48 | hardwire | but do you have chan_zap in your asterisk modules list? |
| 15:53.17 | jakepdev | i did, but i didn't modify that to see the thing work properly |
| 15:53.29 | jakepdev | only thing i did was the modprobe |
| 15:53.35 | hardwire | ok |
| 15:53.46 | adjacent | i have alot of cisco voip gar that i would like to sell off. voip isn't a player in our market. i'm not on the forums enough to know the rules about advertising, either. any recommendations on where to post the equipment for sale? |
| 15:53.59 | jakepdev | join asterisk-biz |
| 15:54.10 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 15:54.21 | adjacent | thanks. |
| 15:54.36 | jaiger | hardwire, the chan_zap will fail to load if the kernel modules aren't loaded |
| 15:54.38 | jakepdev | np |
| 15:54.46 | hardwire | jaiger: ok |
| 15:56.07 | hardwire | jakepdev: so you don't use meetme then on that install |
| 15:56.15 | jakepdev | nope |
| 15:56.34 | hardwire | still crappy |
| 15:56.36 | hardwire | damn |
| 15:56.42 | hardwire | lemme try and answer; wait; |
| 15:57.09 | *** join/#asterisk jontow (jontow@ws.woflsys.net) |
| 15:59.08 | mansol | what variable does the called number get put into? is it $dnis? |
| 15:59.58 | Blackvel | there is also $agi_dnid |
| 16:00.10 | Blackvel | mansol: did you find out what the caller number is? |
| 16:00.50 | jakepdev | blackvel - should be agi_callerid |
| 16:00.54 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 16:00.54 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 16:01.25 | hardwire | jakepdev: whats weird is it actually plays longer.. like its stalling playing back the file |
| 16:01.41 | hardwire | it takes forever to get from the playback to the next hop in the extension |
| 16:02.30 | *** join/#asterisk Genosse_Darklord (~Miranda@p5088A62B.dip0.t-ipconnect.de) |
| 16:03.20 | mansol | Blackvel: i am actually trying to get the called number... have you had much experience with this? |
| 16:04.48 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 16:04.51 | mansol | Blackvel: I am just changing the config now - will try in a second to see if I can get the number |
| 16:04.56 | Blackvel | jakepdev: ah |
| 16:05.08 | Blackvel | mansol: think it is agi_dnid |
| 16:05.15 | Blackvel | or in the dialplan? |
| 16:05.59 | Blackvel | is the caller number : ${CALLERIDNUM} |
| 16:06.18 | Blackvel | the other i dont know |
| 16:06.54 | Blackvel | what is agi_rdnis: unknown |
| 16:07.17 | mansol | blackvel: i read something on www.voip-info.org about rdnis.... |
| 16:07.20 | jakepdev | rdnis i believe is the original called number in a xfer situation |
| 16:07.25 | Blackvel | agi_callerid: "My Name" <the number> |
| 16:07.42 | Blackvel | is very weird for sip devices |
| 16:08.00 | Blackvel | ah |
| 16:08.12 | Blackvel | do you read that usually in your agi application? |
| 16:08.35 | jakepdev | not me |
| 16:08.37 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:09.04 | jakepdev | i use Read to collect the DTMF digits to determine where the call came from |
| 16:09.55 | jakepdev | but that's specific to my app |
| 16:10.11 | mansol | jakepdev: what card are you using? |
| 16:10.30 | mansol | Blackvel: ${CALLERIDNUM} in the dialplan is empty :( |
| 16:10.35 | jakepdev | 2 TE110P cards - might move to the 410 soon |
| 16:11.22 | *** join/#asterisk goldenear (~goldenear@d149.dhcp212-198-168.noos.fr) |
| 16:12.21 | mansol | i am using a TDM04B, 4 fxo modules and I can't seem to get the Called Number - Caller ID comes through though |
| 16:14.14 | drumkilla | there is no such thing |
| 16:14.25 | drumkilla | that's why :) |
| 16:14.37 | tzanger | mansol: you can't get there from here |
| 16:14.42 | tzanger | mansol: POTS doesn't give you that info |
| 16:15.34 | drumkilla | use exten => s ... |
| 16:15.34 | tzanger | mansol: the TDM cards can get whatever a normal phone can. If your normal phone can't get called number, then the TDM can't either |
| 16:15.34 | tzanger | mansol: in fact, I remember speaking to someone (was it you?) about this very thing a few days ago |
| 16:15.42 | onkeltimm | ok, who is running chan_capi on cvs head? |
| 16:17.24 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 16:17.30 | smash- | Hello |
| 16:17.51 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:18.39 | cjk_ | anyone an idea how to create a tiff file compatible to txfax from a pdf |
| 16:19.39 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:19.48 | mansol | tzanger: No - this is the first time I've talked to anyone about it. However, the phone system is sending the Called number as part of the caller id in the format STB+STA where STB is the called number and sta is the caller id |
| 16:20.06 | *** join/#asterisk nwhit (~nwhit@65.107.59.67.ptr.us.xo.net) |
| 16:20.13 | nwhit | ~rfc |
| 16:20.16 | jbot | rumour has it, rfc is Request For Comments. They are available at http://www.rfc-editor.org/, http://www.rfc.net, or ask about rfc 1918 for private address ranges |
| 16:20.16 | mansol | asterisk is seeing the sta portion... |
| 16:20.38 | nwhit | ~rcf |
| 16:20.39 | jbot | somebody said rcf was a very complete firewall script for Linux 2.2. Check it out at: http://jsmoriss.mvlan.net/linux/rcf.html |
| 16:21.01 | *** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net) |
| 16:22.16 | mansol | tzanger: are you certain the card can't support this? it seems strange that it can see the caller id portion of the string but not the first part.... |
| 16:22.30 | tzanger | mansol: it looks like it's getting sent in an incompatible format |
| 16:23.00 | mansol | tzanger: that is very possible - what is the 'correct' format? |
| 16:23.26 | Godsey | is there a way to get asterisk to re-establish a sip connection? |
| 16:23.30 | mansol | tzanger: here is a cut-paste from the phone system manual: Call Forward - All Calls (Ext) C11 + STB + STA |
| 16:23.43 | Godsey | for some reason broadvoice keeps f'n up and starts working if I stop now and re-start asterisk |
| 16:24.11 | tzanger | I don't care what the manual says, only what is actually on the wire... :-) |
| 16:24.28 | mansol | tzanger: ok, how can I see what is on the wire? |
| 16:24.41 | tzanger | you might have to dig into the zaptel code a little there should be some debug info |
| 16:24.50 | tzanger | also are you getting any messages about CID checksum errors or anything? |
| 16:25.16 | mansol | tzanger: no - it happily gets the caller id each time |
| 16:25.29 | nwhit | Does anyone have experience with cisco phones... I am having trouble with them hanging people up with they are on hold for more than 30-45 secs |
| 16:25.52 | mansol | tzanger: unforunately my C skills are limited (I can read the code, barely).... |
| 16:26.01 | *** join/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 16:26.13 | mansol | tzanger: what format does asterisk expect? |
| 16:26.21 | onkeltimm | anyone interested in a howto to get CVS-HEAD, spandsp and chan_capi runnjing? ;) |
| 16:26.27 | tzanger | mansol: it can support several, but I am curious as to if that info is even being sent |
| 16:26.50 | tzanger | I mean is it supposed to come as "123+456" (literal text) and * is only seeing "456" ? |
| 16:27.03 | ClayReiche123 | Does anyone know why I would be getting asterisk@<ip-address> in the From header in my INVITE? |
| 16:27.40 | mansol | I think it should come as C11+0123+0456 but yes, asterisk is only seeing 456... I have a lot of control in this phone system and can actually change the format... |
| 16:28.48 | BoRiS | onkeltimm: Sure :) |
| 16:29.25 | tzanger | mansol: change the "+" to "*" or something, or even 0... see if * is tripping up on the character in the number space |
| 16:30.37 | mansol | tzanger: sorry, the + was only there for fun - the actual dtmf codes are sent as normal (another cut-paste from docs if this helps: The station number will always be 4 digits. The number will be padded out with leading 0’s |
| 16:30.37 | mansol | Example: C21 + STB + TRK -> Trunk number = 02, STB = 234 - DTMF --> 2 1 0 0 0 2 0 2 3 4 |
| 16:31.00 | tzanger | ? |
| 16:31.06 | tzanger | what's 0Rs |
| 16:31.23 | tzanger | ok so it is DTMF CID then |
| 16:31.26 | tzanger | not FSK DTMF |
| 16:31.29 | tzanger | er not FSK CID |
| 16:32.00 | mansol | sorry - i wasn't aware of the difference |
| 16:33.24 | onkeltimm | BoRiS: will update the wiki: http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI with my findings |
| 16:33.46 | onkeltimm | because the patch for CVS-HEAD'there doesn't apply cleanly |
| 16:34.02 | onkeltimm | same with spandsp and cvs head |
| 16:34.16 | jsharp | That CID string looks like Feature Group D with DTMF signalling. |
| 16:34.51 | tzanger | I just wanted to make sure |
| 16:34.56 | ClayReiche123 | http://pastebin.ca/11046 |
| 16:35.14 | tzanger | mansol: is it possible to have the PBX wait a little (500ms) before sending the DTMF? |
| 16:35.19 | *** join/#asterisk HammerToe (~HammerToe@bristol.offices.netsight.co.uk) |
| 16:36.23 | mansol | tzanger: now that I am not sure about - the phone system engineers are asleep now (it's 2am here) but I can ask them in the morning.... i'll have a look to see if it is obvious |
| 16:36.35 | HammerToe | hi all. We are looking to setup an Asterisk PBX at work (in the UK) and I am looking for some advice on phones to use. I am looking for 8 desk phones, pref with headset amps built in. Any ideas> |
| 16:36.40 | tzanger | mansol: or just add a shitload of 0000000s to the front or something |
| 16:36.44 | ClayReiche123 | Does anyone know why would I be getting "From: asterisk@<ip>" in my INVITE message header? |
| 16:36.47 | ClayReiche123 | http://pastebin.ca/11046 |
| 16:36.48 | tzanger | I am trying to see if * is just not seeing it because of timing |
| 16:37.29 | gres | hi all. Can i connect my * to skype? |
| 16:37.35 | mansol | tzanger: sure... unfortunately I'll have to wait until tomorrow to find out if that timing can be changed.... |
| 16:37.41 | tzanger | ok |
| 16:37.51 | mansol | tzanger: Is that format supported? |
| 16:38.01 | tzanger | obviously; you're getting it :-) |
| 16:38.19 | mansol | but only the last portion? |
| 16:39.06 | HammerToe | also, what kind of things can you do with displays on the phones? Is there some kind of protocol to pass messages from the Asterix server to the phones? |
| 16:40.36 | mansol | unfortunately it appears i can only rearrange the format (and remove bits) - not add 00000s to the front :( |
| 16:43.20 | onkeltimm | ok it's there: http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI |
| 16:43.35 | onkeltimm | tell me if this is understandable, as i am no native speaker |
| 16:47.59 | mansol | tzanger: looking back through the logs, sometimes i get 'CallerID returned with error on channel...." |
| 16:49.10 | nwhit | Does anyone have experience with cisco phones... I am having trouble with them hanging people up with they are on hold for more than 30-45 secs |
| 16:50.34 | onkeltimm | okay folks, noiw i got it running i will leave. havaaniceday |
| 16:52.50 | shmaltz | gres, nope |
| 16:54.26 | *** join/#asterisk agave-txlink (phanop@216.81.47.201) |
| 16:54.34 | agave-txlink | what does iaxcompat=yes do, exactly? |
| 16:55.53 | *** join/#asterisk brainchill (~brainchil@bigbrother.kdsi.net) |
| 16:56.10 | brainchill | hey |
| 16:56.19 | ClayReiche123 | Is there anyone here who can help me? |
| 16:56.29 | brainchill | no me ... help me |
| 16:56.56 | *** join/#asterisk [hC] (~hardcore@200.122.157.120) |
| 16:56.56 | brainchill | no me ... help me :-) |
| 16:57.16 | brainchill | what is the problem? |
| 16:57.40 | brainchill | sorry if I missed it I've only been here a second |
| 16:58.44 | ClayReiche123 | brainchill:I'm getting "From: asterisk@<ip>" in my INVITE message header? and I don't know why. http://pastebin.ca/11046 |
| 17:00.15 | brainchill | looking now |
| 17:01.17 | ClayReiche123 | thank you! |
| 17:01.33 | agave-txlink | brainchil, do you know what does iaxcompat=yes do, exactly? |
| 17:02.42 | *** join/#asterisk Ayano (~erik_leee@ppp-70-244-234-29.dsl.spfdmo.swbell.net) |
| 17:04.22 | *** join/#asterisk muntz (~msh@acheron.hsd1.ma.comcast.net) |
| 17:04.46 | bkw_ | agave-txlink, read the sample config |
| 17:04.59 | gambolputty | I am getting a syntax error on the following: $[${ring333} != 7]?103 |
| 17:05.03 | gambolputty | from realtime |
| 17:08.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:08.11 | gambolputty | correction: 333,101,GotoIf($[${ring333} != 7]?103) |
| 17:10.28 | *** join/#asterisk durex (~ironman@200.101.109.138) |
| 17:11.45 | muntz | Can anyone help me with the problem of inbound calls failing to be authenticated? |
| 17:12.23 | muntz | May 5 12:52:00 WARNING[11496]: chan_iax2.c:3849 authenticate_verify: requested inkey 'voicepulse01' for RSA authentication does not exist. |
| 17:12.33 | [hC] | I've got a two tiered infrastructure here. I have a "master pbx" in my office, and some clients each have their own office-specific pbx in their respective offices, with all their local extensions and what not defined. They IAX peer to the master pbx for an overall dial plan. (They dont connect to anyone but me for PSTN access, and from the master pbx, i peer with multiple providers - easier to manage) - Now, what i want to know is, will i always have t |
| 17:12.33 | [hC] | o pass RTP data thru the "master PBX" or is there a way I can make it go direct to the peer, and simply use the master for a dialplan alone? |
| 17:12.37 | [hC] | sorry its so long :( |
| 17:12.53 | muntz | I moved this install from FreeBSD to Debian yesterday. |
| 17:13.11 | muntz | I can dial out just fine. |
| 17:13.22 | agave-txlink | <bkw_> agave-txlink, read the sample config |
| 17:13.28 | agave-txlink | what is a "layered switch" |
| 17:13.32 | agave-txlink | are we talking about an ethernet switch? |
| 17:13.33 | agave-txlink | soft switch? |
| 17:13.37 | agave-txlink | switch made out of tree branch? |
| 17:16.54 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 17:17.58 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 17:19.10 | *** join/#asterisk sean (~sean@iconoclast.caedmon.net) |
| 17:19.38 | wiz8291 | hi |
| 17:19.46 | *** join/#asterisk o-m-a (~ng@40.17-dial.augustakom.net) |
| 17:19.52 | wiz8291 | my caller ID isn't working with sayson (analogue) phones... |
| 17:19.54 | Nuxi | I'm having trouble keeping a Motorola vt1005 registered with *. The ping time from * to the phone is 137ms. Is that too far? |
| 17:19.58 | o-m-a | hello |
| 17:20.28 | wiz8291 | it just doesn't show up :( |
| 17:20.34 | [hC] | Nuxi: you might wqnt to remove qualify-yes from sip.conf, some phones dont like being poked repeatedly |
| 17:20.56 | Juggie | hc, and some phones wont work if you dont :) |
| 17:21.05 | [hC] | Juggie: touche :) |
| 17:21.09 | Nuxi | [hC], the in-office sip phones are fine. |
| 17:21.31 | Juggie | hc, i have a phone which crashes if you dont qualify=yes |
| 17:22.08 | [hC] | All I can tell you is that 134ms is not so high that it would disconnect the phone. |
| 17:22.17 | Juggie | 134ms is fine |
| 17:22.28 | Juggie | humans wont notice till about 400-500ms |
| 17:22.48 | *** join/#asterisk apardo (~apardo@80.26.165.255) |
| 17:23.11 | brainchill | unless there is packet loss in the connection that is |
| 17:23.45 | *** join/#asterisk Moc (~mochouina@h66-201-214-109.gtconnect.net) |
| 17:25.00 | wiz8291 | anyone? |
| 17:27.17 | brainchill | has anyone seen a decent gui client for a w32 desktop to make it easier for users to log in/out of call queues and transfer calls etc |
| 17:27.36 | brainchill | regular users that is not operators per say |
| 17:27.50 | Nugget | "per say" == "per se" |
| 17:28.01 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 17:28.16 | Nugget | http://slacker.com/things/pullet-surprise.php |
| 17:29.38 | brainchill | touche |
| 17:30.08 | *** join/#asterisk adjacent (~scott@nc-65-40-81-64.sta.sprint-hsd.net) |
| 17:31.01 | *** join/#asterisk shaonss (~shaon@acc18-ppp9.hay.dialup.connect.net.au) |
| 17:31.31 | bkw_ | DEV MEETING in 30 min... IAX2/guest@switch-3.asterlink.com/996 |
| 17:32.41 | outtolunc | damn near forgot it was thursday |
| 17:32.41 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l04v-9-80.d4.club-internet.fr) |
| 17:32.49 | opus_ | is broadvoice still down |
| 17:33.07 | wiz8291 | anyone got any ideas on my caller id problem? |
| 17:33.14 | opus_ | whats the problem |
| 17:34.32 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 17:34.42 | syslod | Hello. |
| 17:34.56 | wiz8291 | opus_: analogue extensions don't display the callerid |
| 17:35.06 | wiz8291 | even if i specifically set it with SetCIDNum |
| 17:35.47 | *** join/#asterisk phigan (phigan@ip68-109-169-37.ph.ph.cox.net) |
| 17:36.35 | syslod | Anyone got any pointers on SIP NAT? Particually a Polycom 500 behind nat and Asterisk SIP server on public? |
| 17:37.00 | jsharp | set nat=yes in sip.conf and ride into the sunset. |
| 17:37.31 | syslod | :) I did that but it works only with the first phone. |
| 17:38.03 | syslod | First phone reports on 5060. From there on its gets a weird port and no audio. |
| 17:38.11 | wiz8291 | opus_: any ideas? |
| 17:38.19 | jsharp | Then your NAT is funky. You'll need to set the "local SIP port" and "local RTP port" in the phone to something unique |
| 17:38.45 | syslod | This is a linksys router. |
| 17:39.31 | mansol | what is your caller id problem? |
| 17:39.48 | wiz8291 | basically its not being sent to the phone as far as i can tell |
| 17:39.51 | syslod | jsharp: Does that mean I would have to do that for each phone? |
| 17:39.54 | jsharp | Yeah. |
| 17:39.59 | wiz8291 | the phones are sayson 390s |
| 17:40.12 | syslod | jsharp: Is that the only way? |
| 17:40.16 | wiz8291 | rhino channel bank which supports cid |
| 17:40.21 | wiz8291 | and cid is enabled |
| 17:40.27 | jsharp | The linksys is mapping 5060 on the first phone to 5060 on the public ip...and then it tries to make 5060 on the next phone to 5060 as well. |
| 17:41.16 | kFuQ | wiz8291: why not 'SetCallerID' |
| 17:41.19 | syslod | Ok so is there a way to do that with the ports the poly picks after 5060? |
| 17:41.27 | syslod | Or maybe a firewall that works better? |
| 17:41.51 | jsharp | A firewall that works better. I'm not sure which ones do NAT with port renumbering. |
| 17:42.09 | wiz8291 | kFuQ: it should be set with callerid=xxxx from the zapata.conf anyways |
| 17:42.15 | kFuQ | http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID |
| 17:42.17 | syslod | IS that the feature I need? Port renumbering? |
| 17:42.20 | wiz8291 | i was using setcidnum to try and force just the number |
| 17:42.27 | jsharp | I think they call it PAT. Port Address Translation. |
| 17:42.31 | syslod | So poly with 105060 source gets remapped? |
| 17:42.47 | jsharp | Right. |
| 17:43.02 | jsharp | each 5060 on each phone gets mapped to something unique. |
| 17:43.34 | wiz8291 | kFuQ: out to the sip phones it works fine |
| 17:43.40 | wiz8291 | they display the CID |
| 17:43.40 | syslod | Any firmware hacks for the linksys to do that or sip proxy? Or cheap sip proxy? |
| 17:44.43 | kFuQ | wiz8291: i use SetCallerID in mine and it shows just fine on my analog phones |
| 17:46.50 | wiz8291 | hmmmm |
| 17:46.54 | wiz8291 | this is odd |
| 17:46.57 | wiz8291 | something is breaking it |
| 17:47.43 | muntz | I ned to disable musiconhold. |
| 17:47.45 | muntz | need |
| 17:47.54 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 17:48.07 | *** join/#asterisk shaonss (~shaon@61.68.42.9) |
| 17:53.10 | shaonss | anybody can suggest me which linux will be the best with asterisk? plz |
| 17:53.30 | Nugget | shaonss: it doesn't matter at all |
| 17:53.33 | Nivex | <opinion>Debian</opinion> |
| 17:53.49 | Nugget | asterisk doesn't care which linux you use. hell, asterisk doesn't even care if you use linux at all. |
| 17:54.15 | shaonss | but debian dont have any graphican interface |
| 17:54.22 | Nugget | so use something else. |
| 17:54.23 | *** join/#asterisk |ryan| (leash@c-24-7-159-172.hsd1.ca.comcast.net) |
| 17:54.24 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 17:54.24 | puowvip | except I can't use my BRI card under Windows |
| 17:54.26 | puowvip | heh heh |
| 17:54.28 | puowvip | not like I'd want to |
| 17:54.46 | Nivex | shaonss: you are misinformed. |
| 17:55.14 | dmccollum | Shaonss: CentOS 3.4 is a good choice. It's a free version of Redhat 3.4 |
| 17:55.14 | Nugget | Nivex: if you want to yell at people for not knowing enough about your favorite distro, please do it somewhere else. |
| 17:55.30 | Nugget | linux is linux is linux as far as asterisk is concerned. |
| 17:55.54 | Nivex | on that we can agree :) |
| 17:56.03 | tzanger | Nugget: for the most part, yes, but some distros like to rice-out their particular kernel which causes troubles |
| 17:57.00 | dmccollum | Shaonss: It's ok to use XWindows to setup things, but make sure it's not running once you startup Asterisk |
| 17:57.17 | shaonss | i tried redhat9 and madreke9.1 non of these can detect my sound card :( |
| 17:57.50 | harryvv | shaonss, sounds like you have a issue with the card. |
| 17:57.54 | Nugget | *shrug* asterisk doesn't care about sound cards either. :) |
| 17:58.58 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 18:00.51 | sean | I get this every once in a while.. any idea why? May 4 18:01:04 WARNING[54289]: file.c:550 ast_readaudio_callback: Failed to write frame |
| 18:01.01 | sean | it causes calls to bail/not get answered. |
| 18:01.16 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:05.17 | _Vile | hmm, how would I limit the maximum number of simultaneous calls inbound on a number in the dial plan? |
| 18:05.42 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 18:06.18 | harryvv | in other words you dont want to see all the channels used up by one person? |
| 18:06.35 | harryvv | sean i get that to |
| 18:06.51 | _Vile | I don't want all of the channels to be used up by one inbound 800 # number, yeah. |
| 18:06.54 | harryvv | but if its very once in a while could be a corrupt packet. |
| 18:07.17 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 18:07.43 | queuetue | Are there VOIP providers that suck less? (than BV) |
| 18:07.51 | shmaltz | anybody here from NJ? |
| 18:07.54 | shaonss | dmccollum: 64bit or 32bit? |
| 18:08.14 | ronn | hi guys.. lookin for help on how to install zaprtc |
| 18:08.23 | harryvv | good question I dont know vile. But if say I had advertised a deal of the century house for sale and I got all the incomming calls preventing my other realestate angents from calling out because the channels were used up that would be a problem. |
| 18:08.54 | shmaltz | voip and 911 |
| 18:08.54 | shmaltz | http://mrtmag.com/news/martin_voip_911_050505/ |
| 18:08.59 | _Vile | harry, yep, got a temporary problem and just want to busy out because it's maxing my system... and I'm upgrading tomorrow |
| 18:09.00 | harryvv | Vile that would be a example and may be the same like yours. |
| 18:09.36 | ronn | how do i check if rtc support is compiled into the kernel? |
| 18:09.40 | harryvv | upgrading that system or another one? could put in a temp work station to direct all the calls to vm then switch over. |
| 18:10.15 | shmaltz | interesting: |
| 18:10.16 | shmaltz | http://mrtmag.com/news/intrado_verizon_e911_050305/ |
| 18:12.29 | shmaltz | Yet another one: |
| 18:12.30 | shmaltz | http://www.washingtonpost.com/wp-dyn/content/article/2005/05/05/AR2005050500588.html |
| 18:12.50 | sean | harryvv: hmm. ok, guess it's nothing to worry about, then.. I hate that the call pukes, though |
| 18:13.06 | harryvv | Well one way to go around this is just to have the same termination point in the same 911 calling center area. Dedicate a few pstn lines to 911 calls. |
| 18:13.56 | harryvv | or just sell voip as a long distance service. |
| 18:14.07 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:15.39 | sean | harryvv: the problem is number portability, no? I'm in 514. If I had a vonage box, I could take it with me to my parents place (in 506). There'd be no way for vonage to know where I am. |
| 18:15.55 | bprice20 | I just rewrote my extensions.conf to use a combination of MYSQL for evertyhing |
| 18:16.04 | ronn | how do i check if rtc support is compiled into the kernel? |
| 18:16.04 | bprice20 | and I am happy |
| 18:16.11 | bprice20 | cat /dev/rtc |
| 18:16.15 | harryvv | sean, thats very true |
| 18:16.16 | bprice20 | if its there... it is |
| 18:16.28 | ronn | thanks |
| 18:16.34 | bprice20 | or hell ll /dev/rt? |
| 18:16.36 | harryvv | sean, does the vonage box also allow hookup of pstn line? |
| 18:16.45 | sean | harryvv: no idea. (-: |
| 18:18.25 | harryvv | its great if you are on the road and luckily hotels that do have internet connectivity have voip ports open. |
| 18:20.59 | ronn | bprice: when i do cat /dev/rtc .. it doesn't return |
| 18:21.04 | ronn | just hangs there |
| 18:21.26 | *** part/#asterisk Moc (~mochouina@h66-201-214-109.gtconnect.net) |
| 18:22.35 | *** join/#asterisk Rick_Hunter (~rhunter@03-100.008.popsite.net) |
| 18:24.55 | [hC] | Anyone here do alot of IAX native bridging? Ive got mine working, but obviously the * in the middle terminates the calls in favor of them going direct. This messes with billing data (call duration/etc) - Is there any known way around this? |
| 18:26.24 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 18:28.29 | *** part/#asterisk nixter (~nick@OCI-19-41.OneCall.Net) |
| 18:28.39 | *** join/#asterisk nixter (~nick@OCI-19-41.OneCall.Net) |
| 18:30.39 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 18:31.19 | syle | i'm confused by one of the things i read in the asterisk book |
| 18:31.50 | syle | FXO module supplies you with an FXO port is what i thought, but from boook i gather an FXS module is a FXO port |
| 18:31.52 | Weezey | syle: asterisk and * are the same thing. |
| 18:32.27 | MikeJ[Laptop] | fxo port has fxs SIGNALING |
| 18:32.27 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 18:32.36 | Weezey | FXO module suplies FXS to * |
| 18:33.00 | Weezey | there's a picture... |
| 18:33.30 | Weezey | http://www.digium.com/index.php?menu=fxsvfxo |
| 18:34.17 | syle | that picture isn;t very good |
| 18:34.24 | Weezey | scroll down |
| 18:34.31 | syle | its shows the fxo port port ringing |
| 18:34.39 | syle | but fxo is your outside line |
| 18:34.48 | shido6 | its kind of cool looking, really :) |
| 18:35.09 | Weezey | shido6: it's the best example I've seen. |
| 18:35.12 | syle | yeah the second picture makes sense |
| 18:35.14 | harryvv | Thats a really good flash example. |
| 18:35.15 | *** join/#asterisk jsolares (~jsolares@200.6.219.252) |
| 18:36.30 | Weezey | One time I had a hairy double v. I called the doc, fixed me right up. |
| 18:36.55 | syle | well i got my first digium card, problem is i only have 2 fxo modules, anything i can actually do to test with an outside port or should i wait till fxs modules arrive? |
| 18:36.55 | Weezey | jbot, Weezey is annoying. |
| 18:36.56 | jbot | okay, Weezey |
| 18:36.59 | harryvv | I am trying to find a voip provider that can beat local rates for service into india. Anyone know of any either in india or canada? |
| 18:37.25 | jsolares | syle, you can hook up fxo modules to your telephone line |
| 18:37.34 | Weezey | syle: sure load up xlite and connect to * box. |
| 18:38.32 | Weezey | harryvv: where in india? |
| 18:38.34 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 18:39.08 | harryvv | I think it was bangalor |
| 18:39.18 | syle | how exactly do i dial into * box hehe |
| 18:39.36 | syle | i am still waiting on a voip phone so i can do that actually |
| 18:39.42 | syle | diff number |
| 18:39.47 | wiz8291 | ok |
| 18:39.47 | Weezey | harryvv: 20cents/min here. |
| 18:39.55 | wiz8291 | callerid works on call waiting, but not otherwise :/ |
| 18:40.01 | Weezey | cool |
| 18:40.44 | Weezey | harryvv: that's US$ though. |
| 18:40.51 | harryvv | i know |
| 18:40.57 | syle | hmmm xlite is sip software |
| 18:41.00 | Weezey | yes |
| 18:41.10 | syle | so you mean dialing into the linux box across the room from my windows box? |
| 18:41.14 | Weezey | yes |
| 18:41.17 | syle | ok |
| 18:41.45 | Weezey | harryvv: you know people there? |
| 18:42.05 | Weezey | set up a PRI and *, be our source for India calls. |
| 18:42.12 | harryvv | ohh yea...like 10s of thousands of east indians living within 3 mile radius of me. |
| 18:42.27 | Weezey | Streetsville? |
| 18:43.16 | harryvv | local telus rates into india are 28 cents per min. |
| 18:43.57 | harryvv | weezy, does your company have a presents in india or are you going though a wholesaler |
| 18:44.25 | HeadachesAbound | Anybody here using the C AGI Interface? |
| 18:45.09 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 18:45.27 | durex | folks... |
| 18:45.34 | durex | having problem when trying to install AMP: |
| 18:45.38 | durex | su-2.05b# ./install_amp |
| 18:45.38 | durex | Checking for PEAR DB..FAILED |
| 18:45.38 | durex | [FATAL] PEAR must be installed (requires DB.php). Include path: .: |
| 18:46.03 | *** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 18:46.45 | ariel_ | ~seen jerjer |
| 18:46.50 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 7d 13h 22m 40s ago, saying: 'try ulaw or gsm'. |
| 18:47.30 | dmccollum | Any ideas as to why after about a minute or two of conversation phones on the asterisk side cannot hear the other party, but the other caller can hear? |
| 18:48.14 | jsolares | any nat? |
| 18:48.47 | Weezey | ~seen Weezey |
| 18:48.48 | jbot | weezey is currently on #asterisk (7h 11m 5s). Has said a total of 49 messages. Is idling for 1s |
| 18:49.03 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 18:49.18 | Weezey | dmccollum: what kind of hardware are you using? |
| 18:49.29 | Weezey | I had that happen with Sipura stuff on both ends. |
| 18:49.53 | Weezey | pre-QoS too. |
| 18:50.05 | ariel_ | hello is anyone from nufone on line today? |
| 18:51.02 | harryvv | arial, shodo was just here |
| 18:51.18 | harryvv | why do you ask |
| 18:51.26 | Genosse_Darklord | dmcollum thatzs surely a problem with sip and nat |
| 18:52.40 | ariel_ | I am looking at getting a 800 number we signed up at nufone and put 40 dollars in and when we did there web site to get the 800 number it says none available. |
| 18:52.59 | Weezey | jbot, explain vomit |
| 18:53.03 | ariel_ | Was wanting to know when they are going to be getting them? I need a 800 number voip provider for some testing. |
| 18:53.49 | *** join/#asterisk _asr (asr@pimpbox.latency.net) |
| 18:53.56 | ariel_ | seems very hard to get them via a voip provider. livevoip, and others just don't respond to there emails nor have pepole you can talk too. |
| 18:53.58 | *** join/#asterisk Tili (Tili@202-133-65-177-dialup.sat.net.pk) |
| 18:54.39 | harryvv | arial, I have the same problem with thinktel.ca |
| 18:54.54 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 18:54.56 | AgiNamu | Hello |
| 18:55.09 | AgiNamu | Anyone here know about VoiceXML? just some basics? I want to understand if it sucks as much as I think it does. |
| 18:55.11 | _asr | anybody running asterisk with murf's privacy/screening patch? |
| 18:56.15 | ariel_ | strange that so many sites listed for voip service on the wiki but when it comes down to getting an account with them it's like pulling teeth. |
| 18:56.51 | harryvv | arial, every try iax.cc? |
| 18:56.53 | dmccollum | Weezy: I'm running a dual PIII 866 with 512MB Ram. Two x100p cards. one to Vonage and the other to Bell South. 3 Cisco 7960 phones in the house. |
| 18:56.54 | harryvv | sixtel? |
| 18:57.15 | ariel_ | iax.cc does not have info on 800 numbers I even emailed them. with no responce. |
| 18:57.20 | AgiNamu | I dont put much faith in low-cost no comittment voip providers. |
| 18:57.26 | AgiNamu | at least not many of them. |
| 18:57.27 | ariel_ | sixtel I have not heard of them. |
| 18:57.43 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 18:57.43 | *** mode/#asterisk [+o twisted] by ChanServ |
| 18:57.51 | harryvv | Agi, who do you use |
| 18:57.53 | ariel_ | AgiNamu, yes but I need to get some numbers in for testing to see if we can use there service. |
| 18:57.59 | AgiNamu | so no one knows about VoiceXML... does that mean it sucks? :) |
| 18:58.02 | AgiNamu | A CLEC |
| 18:58.10 | dmccollum | Weezy: The calls that it's happening on are from BellSouth. So the traffic isn't going through a firewall. |
| 18:58.10 | harryvv | ariel_: I use sixtel to call into the states. |
| 18:58.45 | AgiNamu | a CLEC with a few DS3s and a Lucent Compact Switch (formerly Telica) |
| 18:59.38 | AgiNamu | One thing I don't get with VXML.... how the hell you have real dynamic stuff |
| 18:59.43 | ariel_ | funny sixtel and iax.cc come to the same site. |
| 18:59.48 | AgiNamu | VoiceXML doesnt seem to be as good as say FastAGI |
| 19:00.02 | AgiNamu | big mergers in little-VoIP-town? |
| 19:00.15 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-98.autorev.intellispace.net) |
| 19:00.18 | sean | AgiNamu: VXML and AGI are fundamentally different technologies |
| 19:00.41 | AgiNamu | sean, yea, that's what I'm seeing. VXML pretty much imposes some silly concept of how programmers should work |
| 19:00.50 | AgiNamu | and AGI is more of a "ok, here's the damn call, do whatever the hell you want with it" |
| 19:01.04 | AgiNamu | AGI just needs some TTS and ASR extensions |
| 19:01.14 | sean | well, yes. AGI is analogous to CGI. VoiceXML is analogous to (e.g.) a content management system. |
| 19:01.16 | AgiNamu | which im thinking of writing |
| 19:01.28 | sean | AgiNamu: that would be sweet (-; |
| 19:01.43 | AgiNamu | So with VoiceXML, how do I , for instance, create an app that takes an order number, then reads order status? |
| 19:01.52 | jsolares | it's not that complicated, it all depends on what TTS program you are using |
| 19:02.01 | AgiNamu | Do I have to iunvest in an VXML gateway on my premises?? |
| 19:02.05 | jsolares | the AGI part that is |
| 19:02.08 | sean | AgiNamu: yes. |
| 19:02.15 | AgiNamu | and if i host the voice part, then how do they handle it? |
| 19:02.22 | jsolares | long live cepstral |
| 19:02.26 | hardwire | ok |
| 19:02.34 | hardwire | now I can try to tackle the voice prompts issues |
| 19:02.35 | AgiNamu | And how do I do dynamic stuff? all i see is XML.... so I dont know how the hell I'd do anything useful, like say, randomly pick an order |
| 19:02.38 | sean | AgiNamu: their VXML browser gets contents from (e.g.) your webserver. |
| 19:02.38 | hardwire | which are just insane. |
| 19:02.46 | hardwire | anybody ran asterisk on a geode before? |
| 19:02.51 | sean | AgiNamu: generate the XML dynamically. |
| 19:02.53 | hardwire | and had success with gsm playback? |
| 19:02.54 | AgiNamu | So my webserver needs a VoiceXML equivalent of ASP.NET |
| 19:03.03 | AgiNamu | or some framework to generate the XML on the fly... |
| 19:03.04 | AgiNamu | i see... |
| 19:03.13 | sean | AgiNamu: sure.. any language can do it, though |
| 19:03.19 | jaiger | AgiNamu, yes. can't asp.net generate xml? |
| 19:03.23 | AgiNamu | right |
| 19:03.31 | AgiNamu | but you dont want to sit there making XML by yourself |
| 19:03.32 | jaiger | java servlets/jsp can generate whatever you want |
| 19:03.37 | sean | heh. |
| 19:03.37 | AgiNamu | just like no one sits there writing HTML manually |
| 19:03.45 | sean | AgiNamu: that's BS. |
| 19:03.51 | sean | most high-end HTML is built manually. |
| 19:03.56 | AgiNamu | it'd be nice to just do something like "CurrentCall.Playback("my.prompt")" |
| 19:04.09 | jaiger | I'm with sean |
| 19:04.10 | AgiNamu | no, i mean dynamically. i prefer to work with controls |
| 19:04.11 | jsolares | i do that with perl :) |
| 19:04.17 | AgiNamu | like Label1.Visible = true |
| 19:04.29 | sean | I hate that crap. (-: I'm a control freak |
| 19:04.32 | hardwire | http://happy9.bogomip.com/albums/work/tdx/dutchharbor/photoalbum_photo_view?b_start=1 |
| 19:04.37 | hardwire | the solar powered voip system I am setting up |
| 19:04.38 | jaiger | voicexml is a different programming model from agi/fastagi |
| 19:04.43 | AgiNamu | it's equivalent to like manging X drawing routines versus using controls. |
| 19:04.47 | sean | I HAVE started using PEAR::HTML_QuickForm for form processing in PHP, though. |
| 19:05.02 | zno | If I use one of digium's sign span cards, I don't need anything else other than the PRI drop right? |
| 19:05.10 | AgiNamu | Well, I'm thinking of creating SuperFastAGI |
| 19:05.11 | zno | I don't have to buy routers or anything |
| 19:05.15 | AgiNamu | or SuperExtendedFastAGI |
| 19:05.18 | harryvv | hardwire thats really nice. |
| 19:05.24 | AgiNamu | so that it'd beable to handle everything VoiceXML can do |
| 19:05.30 | hardwire | it was fun to build |
| 19:05.31 | hardwire | now I need to get it working |
| 19:05.33 | sean | AGI can already do TTS with Festival.. so I've read |
| 19:05.35 | hardwire | I got 2 voice channels working flawlessly over starband |
| 19:05.37 | hardwire | which is what I needed |
| 19:05.52 | jsolares | what cant regular AGI do that you want to do with SuperExtendedFastAGI? |
| 19:05.53 | AgiNamu | it also seems that any VXML product is stupidly expensive |
| 19:06.11 | AgiNamu | Well, for one, I'd have an authentication framework built in for hosting. Then also allow acccess to the audio streams |
| 19:06.15 | AgiNamu | as well as support ASR and TTS. |
| 19:06.20 | jaiger | hardwire, that looks rather large. can you fit it in the truck? |
| 19:06.26 | *** join/#asterisk ritesh (~ritesh@natint3.juniper.net) |
| 19:06.36 | AgiNamu | Like tihs one hosted VXML place wants $0.18 a MINUTE |
| 19:06.40 | jsolares | you can support tts with regular agi ;) |
| 19:06.40 | sean | AgiNamu: VoiceXML is this layer. You're saying you don't want to handle the low level stuff (draw widgets).. VoiceXML does this. You just create simple XML, and the VXML browser handles it. |
| 19:06.42 | AgiNamu | im thinking... wtf.... |
| 19:06.51 | hardwire | jaiger: sure.. but at this latitude I can't |
| 19:06.56 | *** join/#asterisk santiago (~santiago@63.245.86.187) |
| 19:07.04 | AgiNamu | sean ok i think im getting it more. |
| 19:07.19 | sean | yeah.. there needs to be a good OSS VXML browser that plugs into * |
| 19:07.25 | sean | but I haven't found one. |
| 19:07.29 | AgiNamu | But if it's like web based.... how do you manage state and so on |
| 19:07.39 | AgiNamu | each request for a VXML file woudl be different |
| 19:07.46 | jaiger | AgiNamu, the web has session management |
| 19:07.49 | AgiNamu | does it like use querystring or POST |
| 19:08.09 | AgiNamu | not really. it has cookies, querystrings, and post and your application server creates the illusion of state. |
| 19:08.36 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 19:08.43 | doolph | hi |
| 19:08.50 | _Vile | hmm, how would I limit the maximum number of simultaneous calls inbound on a number in the dial plan? |
| 19:08.54 | jaiger | AgiNamu, at your high "control" level, that's state |
| 19:09.19 | AgiNamu | yea i guess it just depends on the level .On FastAGI the tcp connection is the state |
| 19:09.35 | AgiNamu | So... it'd be feasible to do AGI over XML |
| 19:09.38 | AgiNamu | i mean VXML |
| 19:09.41 | doolph | anyon is still installing asterisk here? |
| 19:09.44 | AgiNamu | vice versa that is |
| 19:09.46 | AgiNamu | VXML over AGI |
| 19:09.49 | *** join/#asterisk onkeltimm (~chatzilla@dsl-082-082-121-245.arcor-ip.net) |
| 19:10.16 | sean | the best way to implement a VoiceXML browser, IMO, is to make it look like a SIP/IAX2 phone |
| 19:10.20 | Weezey | jbot, fatty |
| 19:10.22 | jbot | fatty is probably my ex-girlfriend |
| 19:10.30 | sean | that way, all * has to do is pass it off with a Dial |
| 19:10.36 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 19:10.39 | [hC] | Ive got a context which contains a fallback extension, exten => _. so that i can catch "invalid" extensions that are dialed. This works just fine 99% of the places i use it. However ive got a menu where people are prompted to enter an extension to dial, and i set responsetimeout to 15 seconds, and it always matches _. right away |
| 19:10.42 | Derkommissar | Hello |
| 19:10.45 | AgiNamu | yea, but if I am looking at the entire thing |
| 19:10.51 | [hC] | What is the proper way to 'wait' for an extension to be entered in this case? |
| 19:10.52 | AgiNamu | I want a nice dev API, so say a Mort VB dev could wirte shit |
| 19:10.58 | Derkommissar | Does anyone here know the network desing for net2phone ? |
| 19:11.14 | Derkommissar | does net2phone use asterisk for their systems ? |
| 19:11.18 | HeadachesAbound | hC, WaitForExten i think. |
| 19:11.22 | AgiNamu | are there any good VXML application servers? like as mature as say ASP.NET or JSP ? |
| 19:11.45 | jaiger | AgiNamu, AFAIK you use ASP or JSP |
| 19:11.52 | sean | AgiNamu: browsers. |
| 19:11.57 | [hC] | HeadachesAbound: that doesnt appear to be a command |
| 19:12.04 | AgiNamu | The browser requests the VXML |
| 19:12.06 | jaiger | all the samples I've seen are cold-fusion/asp/jsp |
| 19:12.08 | jsolares | i'll stick with cepstral for tts in agi thank you very much :p |
| 19:12.11 | AgiNamu | But then you have to generate the dynamic BXML |
| 19:12.13 | AgiNamu | VXML |
| 19:12.16 | sean | correct. Apache is a good VXML server. |
| 19:12.22 | Derkommissar | i have a net2phone innomedia and the sip messaging im sniffing looks similar to asterisk... even the structure of the rtp. |
| 19:12.24 | onkeltimm | hC: tried other values for responsetimeout? |
| 19:12.25 | AgiNamu | assuming you want to write xml |
| 19:12.31 | HeadachesAbound | hC, sorry, WaitExten(15). |
| 19:12.39 | [hC] | onkeltimm: nope, but it catches it like, RIGHT away. |
| 19:12.48 | [hC] | Aha |
| 19:12.50 | [hC] | Thanks headaches |
| 19:12.53 | HeadachesAbound | np. |
| 19:13.08 | AgiNamu | but there's no system that just gives you some methods "OnCAllStart()" and then you do stuff like "Playback("foo"); s = WaitForInput(); TTS(DBLookupOrderId(s))"" |
| 19:13.10 | AgiNamu | i.e., code like C |
| 19:13.18 | sean | AgiNamu: vxml.sf.net |
| 19:13.22 | [hC] | oh, waitexten is a macro that is defined on voip-info |
| 19:13.32 | onkeltimm | AgiNamu: Background() ?? |
| 19:13.35 | sean | never used it, but it seems the be a high-level class like you're expecting. |
| 19:13.44 | AgiNamu | onkeltimm, sure, whatever. |
| 19:13.52 | AgiNamu | I jsut think that dealing with a call like a seris of web pages is stupid |
| 19:13.55 | sean | AgiNamu: phpagi.sf.net does these things. |
| 19:14.01 | [hC] | Er |
| 19:14.02 | [hC] | Pardon me |
| 19:14.04 | [hC] | I suck. |
| 19:14.06 | *** part/#asterisk santiago (~santiago@63.245.86.187) |
| 19:14.08 | sean | AgiNamu: if you think it's stupid, then don't use VoiceXML |
| 19:14.11 | AgiNamu | thanks a lot sean |
| 19:14.24 | AgiNamu | No, I think HTML sucks too. but with the right framework, its usable. |
| 19:14.38 | eKo1 | HTML is for markup ONLY. |
| 19:14.43 | sean | heh. and I think HTML frameworks are only good academically. |
| 19:15.05 | AgiNamu | so you dont use any session management or likewise |
| 19:15.06 | jsolares | heh i use $AGI->streamfile(tts_cepstral("foo")); |
| 19:15.10 | AgiNamu | well, that's one way to do it |
| 19:15.12 | sean | reality calls and you have to throw out your little threoretical, "proper" way to do stuff. |
| 19:15.25 | Weezey | stupid reality |
| 19:15.30 | AgiNamu | I use ASP.NET and can't imagine going back to coding in something like PHP or ASP |
| 19:15.32 | [hC] | waitexten worked, sort of.. i guess i should figure out why that catchall was being triggered |
| 19:15.32 | Weezey | jbot, reality is stupid |
| 19:15.33 | jbot | ...but reality is already something else... |
| 19:15.48 | AgiNamu | having an event-driven control-based system is great. and i can always drop down and play with the http stream directly from asp.net if i want to. |
| 19:15.48 | [hC] | It didnt seem to like just waiting on a reponsetimeout. |
| 19:16.05 | AgiNamu | so you gotta use a separate VXML product to recreate the reality of having a single call |
| 19:16.05 | syle | lol aginami, i can;t imagine coding in anything but php or perl |
| 19:16.12 | AgiNamu | I used to do ASP |
| 19:16.19 | AgiNamu | moving to ASP.NET.... I go at least 5 times faster. |
| 19:16.20 | Weezey | I love my php and perl |
| 19:16.24 | syle | same |
| 19:16.25 | Weezey | there isn't anything I can't do with them. |
| 19:16.32 | AgiNamu | there isn't anything i cant do with asp.net |
| 19:16.39 | AgiNamu | you can do anything in anything. |
| 19:16.41 | Weezey | Agi: cron jobs? |
| 19:16.44 | jsolares | ewww asp.net needs to die XD |
| 19:16.45 | AgiNamu | hell, people even use Macromedia |
| 19:16.48 | AgiNamu | Weezy, no problem. |
| 19:16.49 | syle | cron jobs, socket layers? |
| 19:16.51 | Weezey | cool. |
| 19:16.53 | AgiNamu | Sockets, no problem. |
| 19:16.58 | AgiNamu | ASP.NET does everything the CLR can do |
| 19:16.58 | syle | kewl |
| 19:17.01 | AgiNamu | and that's pretty much everything. |
| 19:17.14 | Weezey | hey, if you can do it, more power to you. Most ASP programmers lick nuts though. |
| 19:17.15 | AgiNamu | considering, unlike Java, the CLR can do native calls and even you can run asm inline |
| 19:17.17 | jaiger | AgiNamu, run perl modules? |
| 19:17.20 | AgiNamu | oh yea, like VB programmers. |
| 19:17.23 | AgiNamu | jaiger, sure. |
| 19:17.38 | AgiNamu | VB has some cababilities and you can do stuff in it. but most VB programmers are tards. |
| 19:17.46 | Weezey | exactly |
| 19:17.47 | AgiNamu | and most ASP users are idiots too. |
| 19:17.56 | AgiNamu | that's why MS made VS 2005 ASP.NET SUCK BALLS |
| 19:18.00 | AgiNamu | they took out the whole idea of compiling |
| 19:18.01 | sean | bah.. that goes for any low-entry-curve language |
| 19:18.20 | AgiNamu | cause, as they said, "we have to help users who think HTML is a programming language" <- that's a real quote. |
| 19:18.38 | AgiNamu | nah, in VS2003, you had to understand at least what "build" mean |
| 19:18.40 | AgiNamu | not any more. |
| 19:18.58 | AgiNamu | now its back to script-kiddie-i-know-html-so-i-am-a-programmer land. |
| 19:19.16 | Lee__ | users who think HTML is a programming language are not too bad. It's programmers who think HTML is a programming language |
| 19:19.34 | sean | AgiNamu: http://www.xisc.com/ |
| 19:19.36 | syle | i thought script kiddies were ones that understood abit of shell programming hehe |
| 19:19.39 | jsharp | A friend of mine said it best, if you don't mine a big paste of a quote. |
| 19:19.55 | Lee__ | I think most script kiddies understand how to use wget and perhaps FTP |
| 19:20.06 | sean | if you're into that sort of thing |
| 19:20.24 | jsharp | "Perl can be fast and elegant as much as J2EE can be fast and elegant. In the h |
| 19:20.24 | jsharp | ands of a skilled artisan, it can and does happen; it's just that most of the sh |
| 19:20.24 | jsharp | it out there is built by people who'd be better suited to making sure that my bu |
| 19:20.25 | jsharp | rger is cooked thoroughly." |
| 19:20.30 | jsharp | Dammit. |
| 19:20.44 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 19:20.51 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 19:20.51 | AgiNamu | Lee__, thats what MSFT is catering to now. "programmers" who think HTML is like C, but easier |
| 19:20.53 | syle | i hope they start using perl instead of java on linux on cell phones |
| 19:21.02 | opus_ | perl will just fail without saying anything |
| 19:21.03 | AgiNamu | jsharp, there's some truth to that. |
| 19:21.08 | AgiNamu | justl ike you can write a novel in Notepad |
| 19:21.10 | Lee__ | opus_: perl -w |
| 19:21.12 | sean | "Maybe while the Java world was engaged in talking of high end, super techie stuff, with the words 'enterprise', 'transactions' and 'SOA 'embedded in every sentence, the PHP guys actually went out and created a lot of simple yet very useful software." |
| 19:21.13 | AgiNamu | But Word is going to help you do a much better job. |
| 19:21.17 | sean | (-: |
| 19:21.25 | opus_ | java will throw a million exceptions and prompt end users to read pages of stack traces |
| 19:21.32 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 19:21.34 | AgiNamu | sean, that happens a lot too. but if you take PHP-minded people, and give them java, they can really excel and make fun of BOTH parties. |
| 19:21.54 | cursor | Hello all |
| 19:21.54 | AgiNamu | yea, stack traces. fuck stack traces. I liked ASP. "Error 0x8472372" |
| 19:21.54 | opus_ | PHP .. yeah, right. try to order a multidimensional array |
| 19:21.58 | AgiNamu | at line 20. |
| 19:22.02 | AgiNamu | that's just sooo much better. |
| 19:22.10 | Lee__ | this firewall appliance thing I'm using called m0n0wall claims to have coded the entire FreeeBSD boot script in PHP |
| 19:22.19 | sean | AgiNamu: maybe. I'm PHP minded. I see benefits to certain things in Java. At the same time, though, I can whip out changes in minutes, not hours, with my raw PHP. |
| 19:22.20 | syle | opus: why not? |
| 19:22.21 | Weezey | opus; I dream of that at night. |
| 19:22.31 | AgiNamu | sean, if it takes you hours in java, something is wrong. |
| 19:22.33 | jaiger | I'll take my type-safety and stack traces over "Error 58" any day |
| 19:22.36 | AgiNamu | I make edits WAY faster in ASP.NET than ASP |
| 19:22.43 | AgiNamu | because I have an actual program design |
| 19:22.47 | AgiNamu | not just HTML pages + scritps. |
| 19:22.49 | sean | <shrug> this is a useless argument.. |
| 19:22.53 | AgiNamu | yea |
| 19:22.54 | AgiNamu | it is :P |
| 19:23.06 | AgiNamu | hmm.... it seems like VXML could be conveyed over AGI with no problem. |
| 19:23.07 | sean | we've already got our minds made up.. and they're different schools of thought.. |
| 19:23.12 | AgiNamu | but it doesnt seem like AGI could go over VXML |
| 19:23.13 | opus_ | what would jesus program in? |
| 19:23.13 | syle | there is probably a way to order a multidimensional way easy in php, i would probably do it c style though |
| 19:23.14 | Lee__ | I think we all can agree that MS makes crap and that it's generally understood that they will keep doing it. |
| 19:23.32 | AgiNamu | opus_, Jesus uses Faith::GOD |
| 19:23.35 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 19:23.38 | Weezey | my Dad's the emacs king |
| 19:23.42 | Weezey | I use pico |
| 19:23.43 | jsharp | Jesus turns water into machine code. |
| 19:23.43 | AgiNamu | it's mustard-seed powered. |
| 19:23.46 | AgiNamu | LOL |
| 19:23.49 | syle | weezy! |
| 19:23.55 | AgiNamu | jsharp, that was great |
| 19:24.01 | cursor | Emacs is an interesting approach to an operating system, but it lacks a decent editor. |
| 19:24.11 | opus_ | intellij java ide is the mother fucking bomb |
| 19:24.11 | Lee__ | !! |
| 19:24.12 | syle | weezy: me to man, lately i have been aliasing pico to nano cause all those bastards don;t install pine anymore |
| 19:24.27 | AgiNamu | sean, that PRADO thing looks quite cool. |
| 19:24.31 | opus_ | and ultraedit owns you |
| 19:24.42 | AgiNamu | I like nano. just wish numlock/keypad would work. |
| 19:24.43 | syle | vi owns you :) |
| 19:24.46 | cursor | Is there an Asterisk button graphic I can use |
| 19:24.56 | cursor | for instance, with a link to the main Asterisk website |
| 19:24.57 | eKo1 | no no, vim pwns u |
| 19:25.10 | Weezey | syle: I like pico on nix and I use Programmer's File Editor on my doze box. |
| 19:25.13 | opus_ | cursor yeah, steal the one from nufone.com |
| 19:25.15 | sean | ironically, I use eclipse (and vim) |
| 19:25.19 | AgiNamu | So really, we coudl write a VXML to AGI converter and thus have an Asterisk VXML Gateway. browser. whatever. |
| 19:25.37 | AgiNamu | http://www.asterisk.org/ has images |
| 19:25.40 | eKo1 | I don't see the point of vxml. |
| 19:25.43 | opus_ | voiceXML? i read about that on the onion once |
| 19:25.45 | syle | i use pico no matter what, if on windows box i just securecrt over to the machine |
| 19:25.45 | onkeltimm | Weezey: pico's ideologically wrong. use nano |
| 19:25.56 | cursor | I had a look around asterisk.org (quick look) but couldn't find any |
| 19:25.57 | syle | i;ll switch up to vi on occasion for regex stuff |
| 19:25.59 | AgiNamu | eKo1, I think VXML is silly. reminds me of idiots coming up with stuff for programmers. |
| 19:26.01 | cursor | that "powered by" one will do |
| 19:26.08 | AgiNamu | like a lot of stuff in telephony |
| 19:26.11 | Weezey | onkeltimm: I thought they were the same? |
| 19:26.16 | opus_ | why not wrap the manager APi server in c# tables and just do a .writexml() :) |
| 19:26.20 | AgiNamu | invented by non-coders and sucky. also like banks. |
| 19:26.31 | AgiNamu | opus_, actually, that's whatI am going to do. |
| 19:26.36 | onkeltimm | different license |
| 19:26.40 | syle | i use to hate vi absolutely hate it because it always wordwrapped and nothing you could do about it, vim fixed that |
| 19:26.41 | opus_ | yeah, seems like the best |
| 19:26.44 | *** join/#asterisk jmacz (~jmacz@63.245.86.185) |
| 19:26.49 | opus_ | aginamu - you need any help let me know |
| 19:26.51 | Weezey | onkeltimm: so, other than that they're the same? |
| 19:26.53 | AgiNamu | banks were created by bankers. thats why they have buck-futtingly stupid ideas like "account types" |
| 19:26.59 | AgiNamu | opus_, Well, it's commercially funded. |
| 19:27.02 | onkeltimm | yep. sure ;) |
| 19:27.10 | opus_ | aginamu - i accept money:) |
| 19:27.11 | AgiNamu | so most likely it wouldnt be GPL. |
| 19:27.17 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 19:27.23 | jsharp | bwahhahahahah. |
| 19:27.24 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 19:27.29 | AgiNamu | I like notepad2. |
| 19:27.50 | syle | well weezy: alias pico='nano -w' is fine for me, almost exactly same as pico functions anyways |
| 19:28.00 | cursor | Wunch of Bankers |
| 19:28.03 | eKo1 | I thought banks where created by babylonian prostitutes. |
| 19:28.17 | newmedian | cat > file. A little simplistic, but gets the job done. ;) |
| 19:28.28 | Ayano | can you do ring back tones with asterisk? |
| 19:29.08 | Weezey | jbot, banks |
| 19:29.09 | jbot | banks are what you arrange RAM in, or theives who pretend to help you with your money |
| 19:29.09 | onkeltimm | newmedian: i really use that quite often when I click'ncopy from the web... |
| 19:29.24 | sean | newmedian: bah. cat /dev/random | interpreter | unit_tester; loop |
| 19:30.45 | onkeltimm | sean: million monkey programming? |
| 19:31.02 | tzanger | jbot, botsnack |
| 19:31.02 | jbot | tzanger: aw, gee |
| 19:31.14 | Weezey | jbot is my new best friend |
| 19:31.26 | sean | (-; |
| 19:31.42 | ritesh | can we put astersik + firewall on the same box? or it would be a bad idea? |
| 19:31.54 | eKo1 | Yes you can. |
| 19:31.55 | ritesh | any comments/suggestions |
| 19:32.12 | ritesh | has anyone done this? please send me some pointers/ideas.... |
| 19:32.33 | newmedian | ----> and E=MC2 |
| 19:32.33 | tzanger | ritesh: I do it |
| 19:32.35 | AgiNamu | So, any thoughts on 911 for us small voip companies? |
| 19:32.43 | ionix | outsource it |
| 19:32.46 | ionix | like comwave |
| 19:32.49 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:33.01 | Weezey | Agi: Dial(Zap/g1/911) |
| 19:33.17 | ritesh | tzanger: are there any security issues? would you please elaborate? |
| 19:33.34 | tzanger | ritesh: same security issues as any service running as root and taking packets off the 'net |
| 19:34.01 | *** join/#asterisk rndmtngnt ([U2FsdGVkX@203.13.74.236) |
| 19:35.53 | cursor | So don't run Asterisk as root then |
| 19:35.57 | *** join/#asterisk Marlow (~marlow@159-134-145-35.as1.mvw.galway.eircom.net) |
| 19:36.17 | Weezey | I run asterisk as cursor for safety. |
| 19:36.23 | cursor | :-) |
| 19:36.39 | Marlow | i hate eircom |
| 19:36.43 | AgiNamu | who is comwave? |
| 19:36.57 | AgiNamu | is there anyone I can goto for little money and get E911? |
| 19:37.04 | Weezey | watch out for the IRA |
| 19:37.08 | AgiNamu | Like Intrado is $5K down, $5K minimum a month, 50 cents a Did |
| 19:37.24 | newmedian | Has anyone tried running Asterisk in User Mode Linux, giving slices of modules to UMLs? You could set up a hosting provider offering UML Asterisk - configure your own Asterisk, your own way. And you would get only as many PSTN lines as you need. |
| 19:37.29 | sean | AgiNamu: cell phone with no plan/activation. |
| 19:37.34 | syle | $5K minimum a month ? |
| 19:37.37 | Lee__ | running asterisk not as root is a bit confusing without this reference: http://www.voip-info.org/wiki-Asterisk+non-root |
| 19:37.42 | syle | thats not acceptable |
| 19:37.43 | Weezey | jbot, AgiNamu |
| 19:37.44 | jbot | i guess aginamu is obcessed with 911. |
| 19:37.46 | AgiNamu | syle, yea, so for people with like 5000 DIDs or less, it's a lot. |
| 19:38.04 | *** join/#asterisk SirPrize (~blah@host-84-9-107-20.bulldogdsl.com) |
| 19:38.10 | AgiNamu | Well, im thinking about offering 911 service. |
| 19:38.26 | AgiNamu | Around $850 setup, $50 a month minimum, say, $1 a DID |
| 19:38.34 | Lee__ | newmedian: I was thinking about UML. I already tried within VMware on Windows with a Linux guest and it's hardly functional for RTC issues. |
| 19:38.35 | syle | people can accept minimum purchase minutes but not people telling them how long they have to use them lol |
| 19:38.42 | cursor | Why would someone pay all of that when they can dial 911 for free? |
| 19:39.00 | cursor | or do you have to pay for a police callout in your area? |
| 19:39.07 | harryvv | cursor, you are billed 911 service if you want it or not..its a minimal cost though. |
| 19:39.15 | Lee__ | cellular 911 service has a fee |
| 19:39.23 | SirPrize | How can I configure my Asterisk to automatically figure out that it should connect to anywhere.com when a configured user enters sip:username@anywhere.com as the destination address to be dialled ? |
| 19:39.25 | sean | not if the cell phone is not active |
| 19:39.27 | cursor | Wow - you have to pay to use the emergency services? |
| 19:39.29 | Lee__ | he, sure. |
| 19:39.32 | cursor | That's amazing |
| 19:39.46 | harryvv | cursor, 911 dispatchers dont live with there perents |
| 19:39.48 | Lee__ | cheap city government can't pay for call centers |
| 19:39.48 | harryvv | :) |
| 19:39.53 | sean | even if it's not active to the provider, it should still work. |
| 19:40.02 | cursor | So what does all your tax get spent on then? |
| 19:40.02 | sean | in North America, anyway |
| 19:40.04 | cursor | bombs? |
| 19:40.05 | newmedian | Canada=911 available even if account is inactive; As long as there's a circuit.... |
| 19:40.13 | AgiNamu | VoIP centres can't offer 911 for free. |
| 19:40.17 | Lee__ | cursor: wins the prize!!! |
| 19:40.28 | sean | newmedian: even if there's no dialtone? |
| 19:40.30 | tzanger | newmedian: you sure about that? |
| 19:40.34 | Lee__ | oh, and oil subsidies |
| 19:40.44 | tzanger | newmedian: normally you have battery but no dialtone |
| 19:40.47 | AgiNamu | So would any of you sign up for E911 service at around $900 and $1/did? |
| 19:41.06 | sean | newmedian: for cell phones, this is true... |
| 19:41.35 | Micc | Is anyone here proficient in building iaxclient? |
| 19:41.41 | newmedian | My understanding has been that they are legally mandated to provide 911 service on a circuit, even if the account is inactive. But what they're mandated to do and what they actually do may be two different things. |
| 19:41.45 | Lee__ | Micc: what's yr error? |
| 19:41.55 | Micc | I've built it but I'm having a hard time linking with it. |
| 19:42.00 | AgiNamu | cursor, sure, 911 is paid for by us in the states too. |
| 19:42.06 | AgiNamu | but you, as a VoIP provider, can't edit the ALI |
| 19:42.14 | Lee__ | Micc: linking other source code with the library? |
| 19:42.15 | AgiNamu | and just routing to the correct PSAP aint easy either. |
| 19:42.36 | Micc | Lee__, yes. I get a bunch of unresolved externals. __iob import errors too. |
| 19:42.50 | cursor | I just reserve an analogue line for emergency calls |
| 19:43.27 | Lee__ | Micc: not sure. sounds like it's linking to stuff that doesn't exist. |
| 19:43.37 | cursor | rndmtntgnt: here too - 999 has to work by law |
| 19:43.48 | cursor | even on a mobile phone with no SIM card |
| 19:44.08 | eKo1 | Even on the numeric keypad on the keyboard |
| 19:44.26 | Micc | Lee__, I make a new C# project or C++ .NET project and then link it with the lib and make one call to iaxc_shutdown() |
| 19:44.33 | cursor | All phones have a numeric keypad, so yes |
| 19:44.35 | SirPrize | Can anyone tell me how to configure Asterisk to be able to dial generic SIP addresses of the form sip:someone@somewhere.com by directly connecting to the somewhere.com server? |
| 19:44.46 | Lee__ | Micc: oh, I know nothing about compiling on Windows. sorry. |
| 19:45.02 | sean | SirPrize: just specify the IP in your dialplan |
| 19:45.07 | sean | I do this with my provider |
| 19:45.21 | Micc | Does anyone know anything about building iaxclient on windows? |
| 19:45.34 | cursor | Windows - spit |
| 19:45.35 | sean | Dial(SIP/5147777777@192.168.200.200); |
| 19:45.47 | newmedian | Has someone made a windows dll of iax2/iaxclient? I wouldn't mind invoking it from somewhere. |
| 19:45.48 | SirPrize | sean: but I don't want to connect to just one server. I want to let people connect using any sip address that they have, and which I won't know beforehand what the server would be |
| 19:46.09 | AgiNamu | cursor, great. but as a voip provider, that doesnt help you offer 911 to clients. |
| 19:46.24 | SirPrize | sean: for example, sip:5738234@sipgate.de , sip:2938743@sipgate.co.uk, sip:9349753@gossiptel.com, etc. |
| 19:46.25 | sean | SirPrize: sounds like you want to set them up as friends in you sip.conf |
| 19:46.51 | sean | ah |
| 19:47.03 | SirPrize | sean: but there's no way I would be able to specify them as friends, as I won't know beforehand what sip addresses would be dialed .... :-( |
| 19:47.06 | sean | you could set them up as extensions. |
| 19:47.14 | sean | with wildcards |
| 19:47.26 | sean | what you want to do, though, is allow the CALLER to specify the domain? |
| 19:47.31 | sean | I don't know how to do that (-: |
| 19:47.34 | SirPrize | sean: yes |
| 19:47.37 | SirPrize | oh. :-( |
| 19:47.59 | SirPrize | I've seen that sip addresses of the form I typed above are starting to appear on the internet |
| 19:48.18 | SirPrize | and you just copy/paste into X-lite, and hit dial, and it connects. IT does this via GossipTel for example |
| 19:48.25 | SirPrize | I want to duplicate that functionality in my asterisk server |
| 19:49.00 | file | prefixes |
| 19:49.09 | SirPrize | I mean, if I've configured X-lite to register directly with GossipTel, I can do that. I want to know what I'd need to do in my * if I want to let my X-lite register against my * server |
| 19:49.19 | sean | you'd have to set up a prefix for every possible domain, though |
| 19:49.31 | *** join/#asterisk meshuga (meshuga@c-24-21-94-74.hsd1.wa.comcast.net) |
| 19:49.49 | sean | you could convert the IP to a LONG, and use THAT as a prefix (-; |
| 19:49.58 | file | that's cute |
| 19:50.07 | SirPrize | yeah - someone else also suggested a prefix, having a prefix for each network, but as sean said, there's no way to know all possible prefixes/domains |
| 19:50.16 | SirPrize | anyone can set up a SIP server on their own domain |
| 19:50.31 | SirPrize | I'm sure this must be possible somehow |
| 19:50.52 | SirPrize | Asterisk has the SRV lookup, which is kind of what I'm looking for |
| 19:50.54 | file | I'm afraid not, not without having like SER in front and using it as a proxy |
| 19:51.14 | cursor | sirprise: you want to dial ${EXTEN}@${SIPDOMAIN} |
| 19:51.17 | sean | don't forget that * isn't a SIP phone, it's a PBX that allows you to connect [to] SIP phones |
| 19:51.19 | SirPrize | but even with SRVLookup turned on (which is the default), I haven't been able to set anything up |
| 19:51.28 | file | sipdomain? ugh that's evil |
| 19:51.35 | cursor | it works :-) |
| 19:51.45 | SirPrize | that sounds like it could be misused too :) |
| 19:51.58 | cursor | Just check that SIPDOMAIN isn't one of yours first |
| 19:52.01 | cursor | GotoIf($[${SIPDOMAIN} != ${LOCAL_DOMAIN}] ... |
| 19:52.15 | SirPrize | is ${SIPDOMAIN} populated by * automatically, or isn't the entire address in the ${EXTEN} ? |
| 19:52.16 | cursor | how could it be misused |
| 19:52.17 | Weezey | !=! |
| 19:52.27 | cursor | you restrict its usage to the local users' context |
| 19:52.27 | Weezey | dammit, I wish I knew that existed. |
| 19:52.53 | cursor | if you dial foo@bar.com... |
| 19:53.00 | cursor | EXTEN = foo |
| 19:53.03 | cursor | SIPDOMAIN = bar.com |
| 19:53.09 | SirPrize | ah - brilliant! |
| 19:53.24 | cursor | check it againt your local domain before you use it |
| 19:53.32 | sean | hmm.. good to know |
| 19:54.06 | SirPrize | what extension wildcard should I use to test that in ? :-S |
| 19:54.07 | Nugget | I've got a hacked-up dialplan that lets asterisk act as a sip proxy using ${SIPDOMAIN} |
| 19:54.17 | Nugget | it's functional, but I don't really recommend doin git |
| 19:54.26 | cursor | I see the one on nufone.net |
| 19:54.33 | cursor | but I can't find it on Asterisk.org |
| 19:56.01 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 19:56.09 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 19:56.14 | Nugget | http://slacker.com/~nugget/asterisk7.php is my approach to SIPDOMAIN dialing. |
| 19:56.30 | Nugget | it works, but it sucks. asterisk really makes it difficult to do |
| 19:56.33 | SirPrize | hahaha - Nugget, I just found your site via Google :) |
| 19:56.36 | Nugget | heh |
| 19:56.46 | SirPrize | am reading it now :) |
| 19:57.00 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 19:57.28 | cursor | (nugget) that's very close to what I do |
| 19:57.53 | rvhi | try to compile cdr-mysql, followed http://voip-info.org/wiki-Asterisk+cdr+mysql |
| 19:58.00 | rvhi | it didn't say how to compile |
| 19:58.17 | rvhi | when do i copy the .c file to? |
| 19:58.34 | rvhi | i mean, where... |
| 19:59.08 | eKo1 | I think * detects whether or not you have mysql. |
| 19:59.24 | eKo1 | When you do a 'make' that is. |
| 19:59.40 | rvhi | but where do i put the file in * directory? |
| 20:00.21 | eKo1 | What file? |
| 20:00.31 | newmedian | Is anyone getting Megalink from Bell? Or PRIs from other providers in Canada? Any good/bad comments re: service, price, etc.? |
| 20:00.33 | rvhi | cdr_addon_mysql.c |
| 20:00.46 | newmedian | (in Canada) |
| 20:01.03 | sean | I'm in Canada.. using unlimitel |
| 20:01.11 | sean | good price.. it's just for play, though |
| 20:01.22 | eKo1 | rvhi: read the README and the INSTALL in the asterisk-addons directory. |
| 20:01.23 | Marlow | newmedian : coffee is good |
| 20:01.30 | sean | and it's all DID->SIP ->termination no PRI |
| 20:01.48 | rvhi | eKo1, are you sure? |
| 20:02.00 | eKo1 | About what? |
| 20:02.20 | rvhi | ok, i think i got it |
| 20:02.33 | rvhi | there is no README or INSTALL files |
| 20:02.42 | rvhi | in -addons directory |
| 20:03.14 | rvhi | i just 'make' in that directory |
| 20:03.16 | rvhi | seems working |
| 20:03.17 | rvhi | thx |
| 20:03.42 | cursor | Dammit - almost forgot |
| 20:04.01 | cursor | Time to go and vote these war-mongering muppets out of office |
| 20:05.07 | newmedian | Unlimitel... $2.50 + $0.011/min for a DID, $0.011/min termination. Any minimum requirements (for expenditures, number of units, etc.) Sean? |
| 20:05.26 | sean | newmedian: $50 upfront for usage |
| 20:05.28 | newmedian | Marlow: Blue Mountain blend coffee is... great. :) |
| 20:05.34 | sean | my account balance is $48.39 (-; |
| 20:06.01 | sean | blue mountain == mmmmmmmmm |
| 20:07.11 | sean | if you can afford blue mountain coffee, you won't care about the $50 up front (-; |
| 20:08.32 | newmedian | Sean, I found a place in Toronto that sells a Blue Mountain blend that's as good as prime Blue Mountain. I believe it's just under $40 (cash sales only) for a 5lb bag, if memory serves. |
| 20:09.06 | sean | $40 for a 5lb bag?! CDN!? |
| 20:09.27 | sean | it's ~$35USD/lb retail |
| 20:09.28 | newmedian | hehehhe :) Normally it sells for around $80-110 in other locations. Yes, CDN. |
| 20:09.40 | sean | wow.. |
| 20:09.52 | sean | newmedian: where is it? I'll be in TO later this month, I think |
| 20:11.20 | *** join/#asterisk rcsu (~su@p213.54.79.69.tisdip.tiscali.de) |
| 20:12.36 | newmedian | It is in a place called Kensington Market. The coffee shop is called Casa Acorena (that's close, if not exact spelling), and they keep the big blue/silver foil sealed bags downstairs. |
| 20:13.00 | sean | good to know. Thanks |
| 20:13.04 | sean | <PROTECTED> |
| 20:13.50 | newmedian | I have a good supplier of Keemun tea. If you like Blue Mountain coffee, you'd love Keemun tea. =) |
| 20:14.08 | sean | never had it |
| 20:15.09 | newmedian | It's rich, dark and delicious; Fragrent, powerful black tea. A couple of mugs of Keemun and the world changes, everything is fresh and new, little flowers spring from translucent buds as if from thought bubbles. uh... well... it's very good tea. |
| 20:15.24 | *** part/#asterisk rcsu (~su@p213.54.79.69.tisdip.tiscali.de) |
| 20:15.44 | sean | ha. |
| 20:15.49 | sean | I'll have to try it.. |
| 20:16.11 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 20:17.32 | smash- | hey |
| 20:17.42 | smash- | how can i save a conversation in asterisk to a .wav |
| 20:17.44 | newmedian | Here's a couple of googles: http://www.fmltea.com/tea/Keemun-tea.htm http://www.nashvilletea.com/keemun.html there's probably a dealer of Keemun near you. |
| 20:18.07 | eKo1 | I think sean has got your point already. |
| 20:18.08 | sean | newmedian: I found them already (-: (the pages) |
| 20:18.32 | *** join/#asterisk crich1999 (~crich@86.56.0.135) |
| 20:18.40 | newmedian | Keemun is best when it is fresh. To ensure quality, you should look for the Taylors of Harrogate brand of Keemun Tea. |
| 20:19.26 | smash- | lol |
| 20:19.26 | smash- | why are you guys talkin about tea |
| 20:19.26 | smash- | shit good or what? |
| 20:19.34 | newmedian | It's very good tea. |
| 20:19.45 | bkw_ | is this #asterisk-tearoom? |
| 20:19.47 | bkw_ | NO |
| 20:19.49 | bkw_ | move along!!! |
| 20:20.01 | sean | 16:13 < sean> /p #coffee |
| 20:20.16 | newmedian | you want to strip teas then. hmmph. ok |
| 20:20.22 | tzanger | coffee pee |
| 20:20.23 | tzanger | eww |
| 20:20.25 | Nivex | what, no teabags? |
| 20:20.32 | newmedian | no bags! |
| 20:20.55 | sean | Nivex: nice work.. here we go. |
| 20:21.17 | CoaxD | Happy 4:20 for anyone on east coast! |
| 20:21.40 | tzanger | it's 5pm here, you insensitive clod! |
| 20:21.47 | CoaxD | bastards :/ |
| 20:21.50 | tzanger | :-) |
| 20:21.52 | sean | it's also 05/05/05 not that really has anything to do with 4:20 |
| 20:21.56 | tzanger | nah it's 4:20 here |
| 20:21.59 | smash- | hey |
| 20:22.01 | tzanger | and yes it's my bday |
| 20:22.07 | CoaxD | tzanger: WOOO!!! Happy bday too! |
| 20:22.11 | shido6 | sean Im drivin over |
| 20:22.14 | smash- | anyone know a good pbx gui sound call recorder? |
| 20:22.20 | sean | shido6: ? |
| 20:22.32 | tzanger | CoaxD: danke |
| 20:22.36 | CoaxD | tzanger: Mine was monday :) |
| 20:22.43 | CoaxD | tzanger: I'm old now :/ |
| 20:22.44 | tzanger | happy bday |
| 20:22.46 | tzanger | how old? |
| 20:22.47 | CoaxD | tzanger: THX |
| 20:22.50 | tzanger | I turned 29 today |
| 20:22.50 | CoaxD | tzanger: 29 |
| 20:22.52 | CoaxD | *sigh* |
| 20:22.57 | CoaxD | see??? |
| 20:23.00 | CoaxD | WE'RE ALL OLD! |
| 20:23.16 | sean | file is a young'un |
| 20:23.25 | CoaxD | yes well tzanger and i are all old |
| 20:23.35 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 20:23.42 | newmedian | 29 is old? now I feel old. |
| 20:24.05 | sean | of course you're old. You just spent 10 mins extolling the virtues of tea! (-; |
| 20:24.17 | newmedian | eh, what's that you say? speak up laddie |
| 20:24.28 | sean | <PROTECTED> |
| 20:24.31 | *** join/#asterisk Mw3 (mw3@195.56.193.13) |
| 20:25.40 | newmedian | Well, just because I started with toggle switches for program load, then paper tape, punch cards, and acoustic couplers for 75, 110 and 300 baud, doesn't mean I'm... well... I guess that does mean I'm old. :( |
| 20:26.01 | CoaxD | newmedian: Alright. You're old, sir. |
| 20:26.14 | eKo1 | veteran... |
| 20:26.14 | CoaxD | newmedian: Tho i can say without a lie that i've owned a 110 baud acoustic coupler |
| 20:26.50 | CoaxD | newmedian: Never had that thing working tho. it was about 10 years old when i got it. :/ |
| 20:28.14 | opus_ | hmm.. |
| 20:28.22 | opus_ | lot of goto:'s in asterisk code |
| 20:28.33 | newmedian | I remember transferring programs over the phone line without even an acoustic modem, from the output of a tape loader to the input of a computer across the city. Took many tries for the binary to load across, but it worked. No time sync, not even a UART. |
| 20:29.55 | newmedian | Of course I wore an onion on my belt, which was the style at the time. |
| 20:30.04 | iCEBrkr | Doh! I forget the command to convert .wav files to .gsm |
| 20:30.27 | iCEBrkr | ...and I can't find the little bash script I wrote to zing through an entire directory! |
| 20:31.36 | *** join/#asterisk RazorJack (RazorJack@205.211.152.97) |
| 20:31.38 | sean | heh |
| 20:31.42 | sean | onion belt |
| 20:32.26 | RazorJack | I am trying to set the txgain and rxgain.... and nothing I do seems to change their values. I have edited zapata.conf, reloaded modules, shutdown/startp asterisk.... ztmonitor 1 -v still shows low TX audio level. |
| 20:32.39 | RazorJack | Using zttool, tx/rx levels show 0/0 |
| 20:32.41 | RazorJack | any ideas? |
| 20:32.57 | RazorJack | Using Zaptel 1.0.7 |
| 20:34.41 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 20:35.06 | RazorJack | Its a Wildcard X101P, anybody? |
| 20:35.44 | iCEBrkr | Ahh! sox. |
| 20:36.00 | RazorJack | sox? |
| 20:36.02 | RaYmAn-Bx | isn't there something like ztcfg? Presumably that would set that kind of thing... |
| 20:36.13 | RaYmAn-Bx | (I don't have a zaptel card so i'm guessing) |
| 20:36.26 | RazorJack | RaYmAn-Bx: my /etc/rc.d/init.d/zaptel runs ztcfg |
| 20:36.49 | RazorJack | but ztcfg does seem to open zapata.conf |
| 20:36.56 | RaYmAn-Bx | and it has been run after changing zapata.conf? |
| 20:37.10 | RazorJack | yep |
| 20:37.18 | RaYmAn-Bx | no idea then |
| 20:40.04 | *** join/#asterisk The_X (chris@true.fiberpimp.net) |
| 20:40.22 | The_X | anyone using AgentCallbackLogin? |
| 20:40.32 | The_X | it always asks me to specify the new extension |
| 20:41.31 | *** join/#asterisk pino (~z@host39-28.pool21345.interbusiness.it) |
| 20:43.46 | iCEBrkr | RazorJack: Have you stop/started the module? |
| 20:43.50 | iCEBrkr | after you made your changes? |
| 20:44.07 | iCEBrkr | err, reloaded the module |
| 20:46.34 | RazorJack | icebrkr: yes |
| 20:46.44 | RazorJack | rmmod/modprobe yes |
| 20:47.00 | RazorJack | I edited chan_zap.c, changed txgain = :P |
| 20:47.11 | RazorJack | gonna test that |
| 20:47.34 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 20:47.50 | The_X | is there a way to create a queue so that agents who login get ringed in a specified order? |
| 20:48.09 | RazorJack | The_X: you can set that up with AMP |
| 20:48.11 | *** part/#asterisk KristinG (~KristinG@muppet.geekgirls.us) |
| 20:48.25 | RazorJack | The_X: not sure the syntax in configs though, just saw it when I was looking at AMP |
| 20:48.35 | The_X | as in asterisk management portal? |
| 20:48.49 | RazorJack | yea amp.voxbox.ca |
| 20:48.54 | RazorJack | neat program |
| 20:49.12 | RazorJack | I use it, but hacked their PHP alot to add features I need, simple stuff |
| 20:50.22 | ionix | hey |
| 20:50.29 | ionix | I have spandsp installed on my linux machine |
| 20:50.31 | ionix | but I still get No application 'rxfax' for extension |
| 20:50.35 | ionix | any ideas ? |
| 20:50.40 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 20:50.45 | TUplink | May 5 16:50:04 WARNING[527]: ast_expr.y:475 ast_yyerror: ast_yyerror(): syntax error: parse error; Input: whats that mean |
| 20:50.49 | pino | you must install app_rxfax |
| 20:50.51 | The_X | mrgoku in the house ... ionix |
| 20:50.55 | ionix | The_X :P |
| 20:51.02 | ionix | pino: oh thx |
| 20:53.24 | *** join/#asterisk agallo (~agx@ip-37-53.sn1.eutelia.it) |
| 20:54.03 | agallo | Good night all, i would like to have supervised call transfer; should i use asterisk-cvs or is ok also with Debian/testing 1.0.7 package? |
| 21:02.53 | *** part/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
| 21:03.51 | *** join/#asterisk jeffik (jefik@69.158.20.239) |
| 21:04.02 | agallo | supervised transfer: any wiki, url, howto, suggestions? *g* |
| 21:04.36 | *** join/#asterisk three55ml_laptop (trilluser@cpe-66-25-89-157.satx.res.rr.com) |
| 21:04.52 | The_X | how do we get the queues to actually use the penalty flag? |
| 21:05.29 | three55ml_laptop | Is there a macro or anyone know of anything that does a transfer to an extension asking something like "Press 1 to accept, 2 to reject"? Just don't want to do something that's already been done. |
| 21:07.31 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 21:08.40 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:08.47 | PBXtech | dudes |
| 21:09.29 | *** part/#asterisk agallo (~agx@ip-37-53.sn1.eutelia.it) |
| 21:11.29 | shmaltz | three55ml_laptop, check the wiki and the list |
| 21:11.39 | shmaltz | I have posted one example to the list |
| 21:11.44 | shmaltz | and the wiki has another |
| 21:12.10 | *** join/#asterisk imcdona (~t@c-24-19-99-95.hsd1.wa.comcast.net) |
| 21:12.13 | imcdona | hello all |
| 21:12.39 | imcdona | for the life of me I can't find the paramater to limit the max concurrent calls for a given sip channel |
| 21:12.46 | imcdona | anyone know what it is? |
| 21:13.17 | *** join/#asterisk delphiuk (~Owner@host217-44-239-96.range217-44.btcentralplus.com) |
| 21:18.05 | dsfr | imcdona, I believe the option for that was deprecated. Take a look at the GetGroupCount and GetGroupMatchCount applications. |
| 21:18.49 | delphiuk | I am getting unresolved symbol when trying to modprobe zaptel on a fresh compile of asterisk 1.07 from cvs, any ideas please? |
| 21:19.34 | *** join/#asterisk Auron (Auron@ruben.brandxnet.com) |
| 21:21.33 | CoaxD | delphiuk: make sure symlinks all point to the current kernel source |
| 21:21.40 | CoaxD | delphiuk: i.e. /usr/src/linux and /usr/src/linux-2.4 |
| 21:21.44 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 21:21.45 | CoaxD | (or linux-2.6) |
| 21:22.14 | CoaxD | delphiuk: If it was screwed up, you'll need to recompile/reinstall zaptel |
| 21:22.15 | The_X | AgentCallbackLogin doesn't freaking work |
| 21:22.46 | smash- | Hey |
| 21:22.52 | smash- | do i have to install festival for it to work? |
| 21:23.01 | smash- | i get connect to server failed |
| 21:23.04 | delphiuk | ok, i have the headers pointing to /user/src/linux? i.e. ln -s /usr/src/kernel-headers-2.4.27-2/ /usr/src/linux |
| 21:23.56 | delphiuk | CoaxD: Should it be headers or source that it points to? |
| 21:25.36 | CoaxD | delphiuk: you just need to make sure that those two symlinks are present and correct |
| 21:25.50 | CoaxD | delphiuk: ln -s /usr/src/linux-2.4.6 /usr/src/linux |
| 21:25.52 | CoaxD | delphiuk: ln -s /usr/src/linux-2.4.6 /usr/src/linux-2.4 |
| 21:26.22 | NewSole | hmm... looks like I found a bug with chan_local.c |
| 21:26.29 | CoaxD | delphiuk: if you're running on a redhat/FC kernel, i'm not sure how they package it, but there should be a kernel-source package that contains all the headers you need |
| 21:26.37 | delphiuk | CoaxD: so, it has to be the source that /usr/src/linux points to, and not the headers? |
| 21:26.45 | CoaxD | delphiuk: Yep |
| 21:26.47 | delphiuk | CoaxD: It's on debian sarge |
| 21:26.51 | jmacz | Hi everyone |
| 21:26.54 | CoaxD | delphiuk: Thats how it works |
| 21:27.09 | CoaxD | delphiuk: Ah, yeah, you need to actually install the kernel-source package |
| 21:27.10 | delphiuk | CoaxD: Thanks, I'll re-link and re-try |
| 21:27.22 | delphiuk | CoaxD: Yep, got that installed already |
| 21:27.24 | jmacz | any one with experience with Rhino Channel Banks? |
| 21:27.35 | CoaxD | delphiuk: I roll my own kernels; this is why I haven't ever run into this problem |
| 21:28.08 | CoaxD | delphiuk: Note: I do use the same kernel on many different boxen, but on boxen that i need to compile stuff, i make sure I rsync my build tree |
| 21:28.22 | delphiuk | ok |
| 21:28.31 | CoaxD | delphiuk: Hope you ahve luck! |
| 21:28.46 | delphiuk | Thanks, CoaxD. I am trying agin now |
| 21:29.01 | NewSole | anyone know why chan_local.c is not copying variables from original channels |
| 21:31.00 | AgiNamu | NewSole, I think I saw somehting on -cvs a while ago to make it stop. |
| 21:31.13 | AgiNamu | I think it only copies channels starting with _ or __ now. |
| 21:32.09 | delphiuk | CoaxD: the zaptel stuff won't compile at all now :( |
| 21:32.20 | NewSole | cause my Dial strings are base on accountcodes.... and no Variable Account code are showing up |
| 21:32.26 | *** join/#asterisk pardove (~pardove@195.146.47.206) |
| 21:32.46 | pardove | hi |
| 21:32.57 | jeffik | anybody wored with spectralink? |
| 21:33.16 | pardove | How can i detect a deadlock? |
| 21:33.35 | *** join/#asterisk bjohnson (~bjohnson@ip206-172.dsl.istop.com) |
| 21:35.55 | delphiuk | CoaxD: but it does compile if I re-link to: ln -s /usr/src/kernel-headers-2.4.27-2/ /usr/src/linux |
| 21:38.28 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:39.29 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 21:40.19 | *** join/#asterisk wvbroadband (~User@206.212.51.149) |
| 21:41.42 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:41.55 | PBXtech | dudes |
| 21:42.19 | rvhi | is there a command to show how many pri channels are used? |
| 21:42.26 | PBXtech | show channels |
| 21:42.38 | PBXtech | my bittorrent client keeps freezing |
| 21:42.41 | rvhi | it shows all channels, include sip |
| 21:42.51 | rvhi | i just want to see pri |
| 21:43.18 | PBXtech | zap show channels |
| 21:43.23 | ManxPower | rvhi: there is no command to do that. |
| 21:43.27 | PBXtech | :) |
| 21:43.33 | ManxPower | zap show channels shows CONFIGURED channels |
| 21:43.36 | PBXtech | yea |
| 21:43.39 | ManxPower | not in use channels |
| 21:43.54 | PBXtech | show channels | grep 1234 |
| 21:44.00 | PBXtech | just kiddin |
| 21:44.13 | PBXtech | sill cant get bittorrent workin |
| 21:44.24 | bkw_ | PBXtech, change the ports |
| 21:44.27 | bkw_ | most ISP's block them |
| 21:44.29 | bkw_ | just becase |
| 21:44.36 | PBXtech | its connecting, just freezing for some reason |
| 21:44.37 | bkw_ | I have to put bittorrent on port 5060 |
| 21:44.38 | bkw_ | to get it to work |
| 21:44.44 | bkw_ | thats what it sounds like |
| 21:45.05 | *** join/#asterisk HeadachesAbound (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 21:45.39 | pardove | How can i detect a deadlock? |
| 21:45.58 | PBXtech | bkw are you selling systems or just voice? |
| 21:46.08 | imcdona | Thanks DSFR....I found it here.....http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup |
| 21:47.28 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:47.37 | *** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 21:49.56 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 21:51.17 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:52.18 | *** join/#asterisk darby_t (~tom@dnl249.neoplus.adsl.tpnet.pl) |
| 21:55.20 | ManxPower | pardove: define "deadlock" |
| 21:55.49 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 21:56.21 | ManxPower | ~docs |
| 21:56.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:58.07 | *** part/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 22:01.45 | ManxPower | One of my users had 304,000 messages in his mailbox today |
| 22:02.03 | shido6 | tell his ex wife he's not married to the bizniotch anymore |
| 22:02.05 | shido6 | so stop calling |
| 22:02.46 | *** join/#asterisk sob0l (~sobol_@uo166.internetdsl.tpnet.pl) |
| 22:03.21 | PBXtech | Laughing Out Loud |
| 22:03.27 | terrapen | i'm working on new blocklists |
| 22:03.34 | terrapen | already have lists for most asian countries |
| 22:03.38 | terrapen | and russia |
| 22:03.45 | delphiuk | ManxPower: what is your distro of choice for Asterisk? |
| 22:03.50 | terrapen | i'm going to do an Eastern Bloc list |
| 22:04.00 | terrapen | and a former Soviet republics list |
| 22:04.05 | ManxPower | delphiuk: Mandrake 9.2 |
| 22:04.10 | PBXtech | its like asterisk war dialing to get 300k messages |
| 22:04.12 | terrapen | (these are lists of CIDR blocks for my mail server) |
| 22:04.31 | ManxPower | The MIS manager set up the guy's mail forwarding wrong |
| 22:04.34 | ManxPower | and created an e-mail look |
| 22:04.50 | ManxPower | I think I have the systemwide procmailrc now setup to avoid that |
| 22:05.23 | ManxPower | for fname in `find /home/gtil/.Maildir`; do rm -fv `grep -l "From: MAILER" $fname`; done |
| 22:06.33 | stoyan | terrapen: there are lots of asians and russians in US, too. why don't you block everyone and talk to yourself? |
| 22:06.56 | Nethab | the voices in his head get lonely that's why |
| 22:07.07 | stoyan | :) |
| 22:07.59 | *** join/#asterisk UdontKnow (evaldo@udontknow.staff.freenode) |
| 22:08.46 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 22:09.08 | TUplink | can anyone on FreeWorld dialup call 411? |
| 22:10.58 | smash- | hey |
| 22:11.00 | smash- | can someone help me |
| 22:11.06 | smash- | im trying to setup this recording |
| 22:11.07 | smash- | <PROTECTED> |
| 22:11.20 | smash- | but it says 'wav [' is invalid format |
| 22:11.23 | terrapen | stoyan, it's not about blocking ethnicities |
| 22:11.23 | ManxPower | smash-: look CAREFULLY at "show application record" |
| 22:11.27 | terrapen | its about blocking sources of spam |
| 22:11.38 | terrapen | especially those sources with whom i cannot easily communicate |
| 22:11.41 | smash- | manx were is that file? |
| 22:11.50 | ManxPower | [whatever] means that thing is optional, not to put the [ and ] in |
| 22:11.51 | terrapen | and, before you ask, i block a hell of a lot of US networks, too |
| 22:12.11 | ManxPower | smash-: in the Asterisk CLI. "show application record" Also see the Wiki for EXAMPLES of using Record |
| 22:12.11 | smash- | <PROTECTED> |
| 22:12.28 | ManxPower | ~docs |
| 22:12.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:14.39 | pardove | what is the best way to find out an * box is deadlocked? |
| 22:15.01 | ManxPower | pardove: define "deadlock" |
| 22:15.21 | smash- | thanks manx |
| 22:15.26 | smash- | makes sense now |
| 22:15.28 | smash- | i never read that |
| 22:16.05 | ManxPower | smash-: The Asterisk CLI Terminal is your friend. Hold it. Love it. Send it flowers. |
| 22:16.10 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 22:16.20 | pardove | like safe_asterisk which will start asterisk after a crash, i just want to write a script to detect * deadlocks and restart it |
| 22:16.44 | smash- | manx i know |
| 22:16.46 | smash- | i hate gastman |
| 22:17.08 | ManxPower | pardove: Deallock means what? Can't ping the machine? "asterisk -r" doesn't work? Can't CTRL-ALT-DEL? Can't receive calls? Can't send calls? |
| 22:17.09 | pardove | ManxPower: for me deadlock means a situation that * cant handle any in/outgoing calls |
| 22:17.23 | ManxPower | pardove: I don't know of a way to detect that. |
| 22:17.40 | ManxPower | pardove: it's usually caused by an IRQ conflict or an old Digium card with bugs. |
| 22:17.44 | pardove | thanks ;-( |
| 22:17.49 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 22:18.27 | pardove | ManxPower: no, in some cases a possible bug will cause a deadlock |
| 22:19.33 | pardove | ManxPower: am i right? |
| 22:19.42 | ManxPower | pardove: That's not very common. People here run thousands of calls using asterisk every day and don't experience lockups. |
| 22:19.56 | shido6 | whats the avg size of the digium box when they ship fxs cards? |
| 22:20.02 | shido6 | l w h |
| 22:20.11 | shido6 | i need a ruler |
| 22:20.14 | *** join/#asterisk Rick_Hunter (~rhunter@03-100.008.popsite.net) |
| 22:20.17 | ManxPower | shido6: size of a hardcover text book |
| 22:20.30 | drbrown | I am having trouble getting my variables to work, is there anything special I must do to activate global variables? |
| 22:21.17 | pardove | ManxPower: which REVs of digium cards seems to have problem, i'm using TE405P |
| 22:21.44 | smash- | hey manx do i gotta install Festival? |
| 22:21.46 | smash- | to use it |
| 22:22.37 | ManxPower | pardove: The only cards that I know of with lockup problems are the TDM400P cards. |
| 22:22.48 | ManxPower | The TE cards seem to either totally work or totally NOT work. |
| 22:23.01 | smash- | manx u think Sangoma cards are good? |
| 22:23.21 | ManxPower | pardove: you confirmed by using "cat /proc/interrupts" to make sure the care is not shareing IRQ with any other device" |
| 22:23.23 | ManxPower | smash-: I don't know |
| 22:23.33 | smash- | they got onboard QoS |
| 22:24.20 | ManxPower | QoS does not exist for T-1 Voice |
| 22:24.33 | ManxPower | QoS (as commonly used here) only applies to IP networks |
| 22:24.44 | puzzled | or ATM |
| 22:24.48 | pardove | ManxPower: my box is in APIC mode, i'm a TE405P (puchased 10 months ago) on a dual xeon 3.2GHz, almost 90% of times i have no problem but i get 1-2 deadlock per day |
| 22:25.00 | pardove | i'm=i use |
| 22:25.25 | smash- | i got ip network |
| 22:25.30 | smash- | i got 1 pri and 2 DIA's |
| 22:25.33 | smash- | coming in 3-5 weeks |
| 22:25.37 | ManxPower | pardove: you confirmed by using "cat /proc/interrupts" to make sure the care is not shareing IRQ with any other device" |
| 22:25.45 | ManxPower | care == card |
| 22:26.07 | pardove | ManxPower: as i said my box is in APIC mode so no IRQ sharing at all |
| 22:26.23 | ManxPower | pardove: Um, I've seen boxes in APIC mode share IRQs. |
| 22:26.35 | ManxPower | Some BIOSs can be amazingly brain dead. |
| 22:28.01 | pardove | ManxPower: 52: 2151348 37341947 0 40626435 IO-APIC-level t4xxp |
| 22:28.13 | pardove | ManxPower: no other resources use IRQ 52 |
| 22:28.17 | ManxPower | pardove: contact Digium support. |
| 22:29.25 | pardove | ManxPower: I'm using SuperMicro X5DP8-G2, any known issue with this MB? |
| 22:30.40 | ManxPower | LOL! |
| 22:30.53 | *** part/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 22:30.55 | smash- | ~docs |
| 22:30.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:30.57 | pardove | ManxPower: What's your idea about Sangoma A104 is it better to use (i prefer to support Digium) |
| 22:31.12 | ManxPower | pardove: We had three supermicro boards. caused massive PRI Abort messages, the second was DoA, the 3rd died after 24 hours. |
| 22:31.20 | ManxPower | pardove: I have never used Sagnoma |
| 22:31.40 | smash- | www.sangoma.com |
| 22:31.47 | smash- | not sag |
| 22:31.48 | smash- | lol |
| 22:31.54 | smash- | they expensive |
| 22:31.54 | pardove | ManxPower: wow! my box is working about 10 months |
| 22:32.01 | smash- | it says it has perfect irq sharing abilities |
| 22:32.17 | smash- | auto 3.3 v or 5v |
| 22:32.21 | shmaltz | pardove, what distro? |
| 22:32.27 | smash- | <PROTECTED> |
| 22:32.28 | pardove | ManxPower: so, what is your dual xeon MB suggestion |
| 22:32.44 | ManxPower | Just remember that if you use a non-Digium card very few people here will have any experience with non-Digium hardware |
| 22:32.48 | smash- | i can let u know how it does in 5 weeks |
| 22:32.53 | pardove | shmaltz: Fedora 2 with fresh 2.6.9 kernel |
| 22:33.01 | smash- | yeah manx thats true |
| 22:33.08 | smash- | sangoma hard to get ahold of too. |
| 22:33.13 | ManxPower | pardove: Intel has two motherboards that are Dual Xeon. We used one of them successfully, but I don't recall exactly which one. |
| 22:33.22 | shmaltz | pardove, any zap hardware? |
| 22:33.27 | shmaltz | pardove, stable or HEAD? |
| 22:33.31 | pardove | shmaltz: TE405P |
| 22:33.43 | ManxPower | pardove: contact Digium. They are (supposed to be) interested in fixing lockups of that kind. |
| 22:33.43 | pardove | shmaltz: HEAD |
| 22:33.53 | shmaltz | nice |
| 22:34.12 | shmaltz | my electricity goes out more often than that :) |
| 22:34.17 | pardove | ManxPower: thanks, do they have any support via IRC? |
| 22:34.30 | eKo1 | eh, no |
| 22:34.48 | ManxPower | pardove: Not officially |
| 22:34.52 | ManxPower | support@digium.com |
| 22:35.08 | ManxPower | be persistant. Their first level techs seem to only want to suggest the most common fixes. |
| 22:36.19 | *** join/#asterisk jsolares (~jsolares@200.6.213.211) |
| 22:36.37 | delphiuk | ManxPower: you seem to be a wisdom of knowledge :) would you be able to tell me whether /usr/src/linux should point to either the kernel sources *or* the kernel headers? |
| 22:37.02 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:37.09 | ManxPower | delphiuk: I dunno. I can build zaptel when I have glibc-headers installed AND kernel-source installed. |
| 22:38.03 | eKo1 | /usr/src/linux points to the sources usually |
| 22:39.27 | opus_ | delphiuk - yeah i think you can get away with it |
| 22:39.40 | Inv_arp | delphiuk: /usr/src/linux sources /usr/include headers |
| 22:39.59 | delphiuk | ManxPower: I can compile when it is pointing to the kernel headers *but* I can't mocprobe |
| 22:43.41 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 22:44.20 | *** part/#asterisk SirPrize (~blah@host-84-9-107-20.bulldogdsl.com) |
| 22:44.26 | pardove | ManxPower: before contacting digium support, i get these log messeges at deadlock time: Avoided deadlock for 'Zap/15-1', 10 retries! - Unable to allocate channel structure - Cannot allocate new structure on channel 3 |
| 22:45.35 | Romik | hello, i have following errors http://pastebin.ca/11080 on asterisk 1.07 - anybody can advice? |
| 22:46.18 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 22:46.50 | pardove | Romik: can reproduce it on HEAD? |
| 22:49.03 | Romik | pardove: it happend frequantly....i abit affraid to replace 1.07 to HEAD....on production pbx |
| 22:49.08 | *** part/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 22:49.47 | Romik | pardove: also i receive such message more freqently May 5 18:46:45 NOTICE[31878]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/livevoip/20 of format speex since our native format has changed to ulaw |
| 22:50.10 | smash- | hey pardove how many circuts do you have to need a dual xeon? |
| 22:51.20 | pardove | smash-: depends on many parameters: such as type of codecs, trancoding, ... |
| 22:51.52 | smash- | u doing any trancoding? |
| 22:52.03 | smash- | im just wondering |
| 22:52.12 | *** join/#asterisk adjacent (~scott@64.203.220.105) |
| 22:52.50 | pardove | smash-: not at all, but i have 50-100 concurrent calls with about 1000 sip extensions |
| 22:52.59 | smash- | o wow |
| 22:53.03 | smash- | ok 1000 sip extensions |
| 22:53.04 | pardove | smash-: and 96 PSTN lines |
| 22:53.13 | smash- | whats pstn mean again |
| 22:53.25 | shmaltz | ~pstn |
| 22:53.26 | jbot | somebody said pstn was Public Switched Telephone Network |
| 22:53.41 | Romik | public - plain |
| 22:53.42 | smash- | are you doing any voip? |
| 22:53.44 | shmaltz | pardove, you using BRI? |
| 22:53.49 | pardove | smash-: Telco lines connected to FXO cards on ChannelBank |
| 22:54.13 | shmaltz | pardove, what type of channel banks? |
| 22:54.25 | pardove | TA750 |
| 22:54.32 | shmaltz | who makes them? |
| 22:54.51 | smash- | i need a like 4-10 port channel bank |
| 22:54.55 | smash- | or should i just use voice modems? |
| 22:54.56 | pardove | shmaltz: Adtran TotalAcces 750, but i don't recommend it |
| 22:55.02 | shmaltz | why not? |
| 22:55.13 | Romik | i just order today 5 x 24FXS of Zhones. |
| 22:55.14 | shmaltz | you would go with CAC rather? |
| 22:55.14 | pardove | shmaltz: echo, echo, echo, |
| 22:55.33 | pardove | shmaltz: CarrierAccess is much better |
| 22:55.38 | shmaltz | i c |
| 22:55.41 | Romik | CAC has AGC? |
| 22:55.56 | shmaltz | whats AGC? |
| 22:56.08 | Romik | AGC = automatic gain control |
| 22:56.16 | pardove | Romik: no, but the have auto impedence feature which will reduce echo |
| 22:56.58 | pardove | Romik: maybe AGC means that ! |
| 22:57.32 | Romik | pardove: with zhones in office we have no any echo, but to other destination we have some times....need find right values for rxgain and txgain. |
| 22:58.28 | pardove | Romik: 1st try to set echocancel=yes on zapata.conf |
| 22:59.06 | Romik | pardove: we have it. but still problem when we call to far destinations...via voip, when we call via PRI - no such problem. |
| 22:59.06 | smash- | hey anyone have a suggestion as to what i should do to create 4-8 pot lines off asterisk |
| 22:59.17 | pardove | Romik: also echocancelwhenbridged=yes |
| 23:00.26 | Romik | we have 3 pbx...new york, middle east, carribians - asterisk rocks! |
| 23:01.11 | pardove | Romik: u're doing voip? |
| 23:01.44 | Romik | pardove: yes |
| 23:01.56 | *** join/#asterisk aspworld (~aspworld@209.91.159.221) |
| 23:02.16 | *** part/#asterisk aspworld (~aspworld@209.91.159.221) |
| 23:02.28 | smash- | hey what's a channel bank that someone would represent? |
| 23:02.35 | smash- | romik asterisk does rock |
| 23:02.45 | smash- | i run it on a sony x505 laptop ontop of windows |
| 23:02.51 | Romik | pardove: 2 office connected , and dispatch all intl calls via new york pbx...and incomming calls come to new pbx |
| 23:02.51 | smash- | uplinked to main server |
| 23:02.54 | smash- | its super sweet |
| 23:03.05 | smash- | runs great on 1.1ghz |
| 23:03.31 | pardove | Romik: which MBs u use? |
| 23:03.56 | Romik | what is MB? |
| 23:04.02 | smash- | motherboard |
| 23:04.07 | pardove | Romik: MainBoard |
| 23:04.27 | pardove | ormotherboard |
| 23:04.29 | Romik | pardove: we connect channel banks (zhones) to digium quad card |
| 23:04.32 | pardove | or motherboard |
| 23:04.37 | Nethab | monkey balls |
| 23:04.50 | smash- | romik u like the zhone channel bank? |
| 23:05.16 | Romik | pardove: motherboards i even do not know which...something generic in once place, and DELL dual 800mhz something in other.. |
| 23:05.17 | pardove | pardove: i'm going to test pleiades CBs |
| 23:05.35 | Romik | smash: i loves them, becouse of price;) |
| 23:05.44 | *** join/#asterisk md` (illuminat@md.boerde.de) |
| 23:05.46 | smash- | so there a good buy? |
| 23:05.54 | md` | hi |
| 23:05.56 | smash- | i need a small one to hook up fax machines |
| 23:06.12 | Romik | smash: you also need T1 card...that is more expensive |
| 23:06.23 | smash- | i have it |
| 23:06.29 | smash- | i have a sangoma qaud card |
| 23:06.32 | smash- | quad |
| 23:06.40 | smash- | was like 1800 dollars |
| 23:06.41 | Romik | smash: it good? |
| 23:06.44 | smash- | dun know |
| 23:06.50 | smash- | pri getting install end month |
| 23:06.51 | Romik | smash: why not digium? |
| 23:06.53 | pardove | smash-: is it good? |
| 23:07.02 | smash- | um my friend richmedium |
| 23:07.07 | smash- | told me it was best they make |
| 23:07.11 | smash- | and he has 6 call centers on it |
| 23:07.25 | smash- | and money wasnt really an issue |
| 23:07.29 | Romik | smash: serius! |
| 23:07.35 | pardove | smash-: are u using it now? any problem? |
| 23:07.37 | smash- | we spent like 10k on snom phones |
| 23:07.43 | smash- | pardove no not yet |
| 23:07.47 | smash- | its in its blue box still |
| 23:07.48 | Romik | smash: for me money also not a issue, but i like to save...in general |
| 23:07.57 | smash- | yeah so does my boss |
| 23:08.08 | smash- | i have to make the rj45 cables for softphone computers |
| 23:08.14 | smash- | to save couple hundred bucks |
| 23:08.15 | smash- | on cable Lol |
| 23:08.17 | pardove | Romik: why dont u use Mediatrix 1124? |
| 23:08.29 | Romik | pardove: what is this? |
| 23:08.34 | smash- | and the snom phones i like alot |
| 23:08.54 | pardove | Romik: 24fxs port sip gateway |
| 23:08.56 | smash- | ill let u guys know about the sangoma card though next month when everything is setup |
| 23:09.11 | Romik | pardove: nice... |
| 23:09.15 | TUplink | what would be a good extention to use for an admin Extention |
| 23:09.16 | smash- | im prob gonna build a channel bank |
| 23:09.35 | Romik | pardove: what price of such bank? |
| 23:09.39 | md` | what would be the easiest way to configure asterisk? |
| 23:09.49 | smash- | sip |
| 23:09.56 | smash- | md thats a broad question |
| 23:10.01 | md` | i know :/ |
| 23:10.05 | smash- | what kinda phones u gonna use |
| 23:10.09 | md` | isdn |
| 23:10.16 | smash- | hrmm... |
| 23:10.16 | pardove | Romik: not sure ~2900$ |
| 23:10.20 | md` | or |
| 23:10.25 | md` | well |
| 23:10.27 | smash- | how many lines |
| 23:10.36 | smash- | u gonna use iax or sip ohones |
| 23:10.36 | md` | just one line i think |
| 23:10.37 | Romik | pardove: expensive shit... |
| 23:10.37 | smash- | phones |
| 23:10.38 | pardove | Romik: but remember that u dont need any T1/E1 port at all |
| 23:10.50 | Romik | pardove: i undestand... |
| 23:11.04 | md` | well i think i misunderstood you, mainly i wanted to use a avm B1 |
| 23:11.09 | pardove | Romik: and also additional * box for them |
| 23:11.18 | smash- | how many lines is a BRI? |
| 23:11.19 | smash- | 4? |
| 23:11.32 | pardove | 2 |
| 23:11.44 | Romik | pardove: i pay for the card $1500-1600, + 4x T1 channel banks for 4x160! it 96 lines...for $2200 |
| 23:12.07 | smash- | romik y not use soft phones |
| 23:12.08 | pardove | zhone CBs are 160$ |
| 23:12.11 | smash- | and buy a switch? |
| 23:12.25 | smash- | pardove how many lines are on the zhone cb |
| 23:12.43 | smash- | for that price. |
| 23:12.52 | Romik | smash: a bit soft phones..firefly...but need to use headset for them...i do not like |
| 23:12.55 | pardove | smash-: Romik has them |
| 23:13.14 | smash- | o |
| 23:13.14 | Romik | smash: 24 lines |
| 23:13.21 | smash- | u cant transfer calls through cb can u>? |
| 23:13.29 | md` | and i more meant what would be the easiest way, as in not manually editing config files but using some kind of configurator... |
| 23:13.34 | pardove | Romik: just FXS version? |
| 23:13.48 | Romik | pardove: i have digium 1 line work as E1 PRI, and 3 lines as 24FXS each |
| 23:14.08 | Romik | pardove: now i adding additional card + 4 x 24fxs |
| 23:14.14 | pardove | Romik: nice |
| 23:14.30 | smash- | romik all those lines move via voip? |
| 23:14.45 | pardove | Romik: where do u buy them? |
| 23:14.50 | smash- | er fxs=outgoing or incoming? |
| 23:14.53 | Romik | smash: local calls moves via PRI, all international calls via voip |
| 23:15.19 | Romik | pardove: on ebya |
| 23:15.21 | Romik | ebay |
| 23:15.23 | smash- | i c |
| 23:15.37 | smash- | ur company saving money Romik? |
| 23:15.38 | pardove | Romik: so they are not new? |
| 23:15.55 | Romik | pardove: they new, but comes with strange labels....like OEM |
| 23:16.23 | smash- | later gotta go home |
| 23:16.29 | pardove | Romik: Zhone real price is more than 1200$, so they won't be new |
| 23:16.45 | Romik | pardove: they discontinued products... |
| 23:17.04 | pardove | Romik: so now who make them? |
| 23:17.35 | Romik | pardove: they old from some warehouse... |
| 23:17.52 | md` | smash-: any hints? |
| 23:18.25 | Romik | pardove: only need to prepare for them cable 50 pin to patchpanel. |
| 23:18.30 | *** join/#asterisk doug` (~seebe@pool-71-104-35-186.lsanca.dsl-w.verizon.net) |
| 23:19.01 | doug` | How can i edit the web meetme control to show Caller ID NUmber and not name? |
| 23:19.07 | doug` | i cant find it in the php source |
| 23:19.26 | doug` | i have the old one where it shows the name in quotes "NuFone Call" then ID, channel, conf number, and mode |
| 23:30.55 | sivana | heh |
| 23:31.03 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 23:35.30 | cjk_ | anyone an idea how to create a tiff file compatible to txfax from a pdf |
| 23:36.17 | Mavvie | tiff2ps, ps2pdf |
| 23:37.24 | cjk_ | Mavvie: thats working great for rxfax. i need the other way round |
| 23:37.37 | Mavvie | oh wait. |
| 23:38.08 | Mavvie | pdf2ps, pstopnm, pnmtotiff |
| 23:38.19 | Mavvie | doesn't always work fine btw. |
| 23:38.31 | Mavvie | there are some postscript files which pstopnm can't convert. |
| 23:38.50 | *** join/#asterisk pbx123 (~ponaps@210.213.216.64) |
| 23:38.57 | JunK-Y | Mavvie: where can we find pdf2ps ? no deb package? |
| 23:39.54 | Mavvie | [~] edwin@k7>pkg_which `which pdf2ps` |
| 23:39.55 | Mavvie | ghostscript-afpl-8.14_6,1 |
| 23:43.31 | JunK-Y | Mavvie: thx |
| 23:43.45 | JunK-Y | i guess i can create it via a simple .txt too? |
| 23:43.54 | jayk2 | i'm trying to set up a 7960 to retrieve voicemail when hitting the Messages button |
| 23:44.15 | jayk2 | i added messages_uri to both SIPDefault.cnf and SIP<mac>.cnf and rebooted, but when hitting the button, it does nothing |
| 23:44.25 | jayk2 | i can get to the voicemail system by entering the voicemail number, however. |
| 23:44.34 | shido6 | messages_uri |
| 23:44.45 | shido6 | jayk2, look in your sip<macaddress>.cnf |
| 23:44.58 | shido6 | crap you're gonna make me post to pastebin arent ya |
| 23:45.07 | jayk2 | what do i look for? |
| 23:45.14 | jayk2 | i added messages_uri: "2500" |
| 23:45.18 | jayk2 | 2500 is the voicemail system |
| 23:45.24 | shido6 | no |
| 23:45.30 | shido6 | well you COULD do it that way |
| 23:45.31 | jayk2 | oh. |
| 23:45.32 | shido6 | but think further |
| 23:45.38 | shido6 | whats the extension for that phone? |
| 23:45.44 | jayk2 | 120 |
| 23:45.50 | jayk2 | do i put that there? |
| 23:45.57 | shido6 | ok... and wouldnt it be cooler if you could dial 25120 |
| 23:46.05 | shido6 | and have the vmail system just ask you for a pword? |
| 23:46.17 | shido6 | well that would require you to rethink your vmail portion of your dialplan , right? |
| 23:46.22 | shido6 | how do you think that would look... |
| 23:46.35 | jayk2 | it asks me for a password already when i hit 2500 |
| 23:46.40 | PatrickDK | heh, or just do it so if you dial your extention from your phone, it gives you voicemail |
| 23:46.58 | doug` | i have the old one where it shows the name in quotes "NuFone Call" then ID, channel, conf number, and mode |
| 23:46.59 | jayk2 | i catch the extension using CALLERID in the extensions.conf |
| 23:47.00 | doug` | How can i edit the web meetme control to show Caller ID NUmber and not name? |
| 23:47.02 | doug` | i cant find it in the php source |
| 23:47.22 | shido6 | http://pastebin.ca/11087 |
| 23:47.31 | jayk2 | but still |
| 23:47.42 | jayk2 | if i have messages_uri: 2500, shouldnt it dial 2500 when i hit the messages button? |
| 23:47.45 | shido6 | no |
| 23:47.53 | shido6 | the message_ur:25120 |
| 23:48.01 | jayk2 | oh. |
| 23:48.06 | shido6 | set the messages_uri: to 25120 |
| 23:48.09 | shido6 | get it? |
| 23:48.12 | shido6 | so now when u push the button |
| 23:48.13 | shido6 | BAM |
| 23:48.16 | shido6 | "password?" |
| 23:48.16 | jayk2 | let me try that |
| 23:48.30 | shido6 | or you could do it PatrickDK 's way and have the callerid verify the user |
| 23:48.33 | shido6 | but i dont suggest that |
| 23:48.45 | PatrickDK | well, they still enter the password |
| 23:48.53 | PatrickDK | but I like to give my users many options |
| 23:48.59 | shido6 | I thought you wanted the cid to verify without pword |
| 23:49.00 | PatrickDK | so they can do it the way they are used to doing it |
| 23:49.01 | shido6 | ok, gotchya |
| 23:49.56 | jayk2 | ok |
| 23:50.00 | jayk2 | one other thing |
| 23:50.01 | shido6 | work? |
| 23:50.19 | shido6 | fire away, jayk2 because soon the nicotine wears off and I'll be /away again |
| 23:50.21 | jayk2 | when my phone boots up |
| 23:50.27 | shido6 | yesss? |
| 23:50.32 | jayk2 | it sits on "Configuring VLAN" for about 2 minutes |
| 23:50.44 | jayk2 | im running 7.4 on the phone |
| 23:50.50 | shido6 | heheeh |
| 23:50.53 | shido6 | 2 minutes is bad |
| 23:50.59 | shido6 | it does take a while tho |
| 23:51.04 | shido6 | let me reboot mine and find out what ti does |
| 23:51.16 | shido6 | configuring vlan |
| 23:51.24 | shido6 | configuring ip |
| 23:51.31 | jayk2 | mine is still at configuring vlan |
| 23:51.33 | shido6 | requesting configuraton |
| 23:51.36 | jayk2 | its connected to a 2924XL |
| 23:51.48 | shido6 | (my tftp server is down so lets see how long it will take to bump into default) |
| 23:52.29 | jayk2 | messages button still doesnt work |
| 23:52.43 | shido6 | okie dokie you screwed up somewhere :) |
| 23:52.56 | shido6 | pastebin.ca your dialplan, sip.conf and vocemail.conf |
| 23:52.56 | jayk2 | ok |
| 23:52.59 | shido6 | ok phone is done |
| 23:53.01 | shido6 | back up |
| 23:53.07 | shido6 | less than a couple minutes |
| 23:53.41 | *** part/#asterisk delphiuk (~Owner@host217-44-239-96.range217-44.btcentralplus.com) |
| 23:54.46 | shido6 | Cisco’s Catalyst 2924 XL 10/100 autosensing Fast Ethernet switch |
| 23:55.00 | jayk2 | yeah |
| 23:55.09 | jayk2 | im not too concerned about the Configuring VLAN part |
| 23:55.30 | jayk2 | at least not yet |
| 23:55.32 | jayk2 | :) |
| 23:56.12 | *** join/#asterisk jackfiber (cico@213.217.38.55) |
| 23:56.13 | shido6 | well u sorta need to |
| 23:56.18 | shido6 | it has to go through the step |
| 23:56.25 | jackfiber | hello, anyone here is using Mediaproxy? |
| 23:56.58 | cjk_ | Mavvie: i found many ways to create the tiff file. but it seems to be not compatible with txfax |
| 23:57.33 | JunK-Y | cjk_: how do ya use txfax? with caller? |
| 23:57.34 | shido6 | ok |
| 23:57.39 | shido6 | who sent me their dialplan |
| 23:57.44 | jackfiber | cjk_> soft-switch spandp is working with Asterisk 1.x ? |
| 23:57.53 | jackfiber | spandsp I meant |
| 23:57.58 | jayk2 | i sent you sip.conf, extensions.conf, and voicemail.conf |
| 23:58.01 | jayk2 | i dont have a dialplan set up yet |
| 23:59.27 | cjk_ | JunK-Y: txfax is working for sure. if i send a file i recieved with rxfax it works |
| 23:59.31 | twisted[work] | extensions.conf == dialplan |
| 23:59.38 | cjk_ | jackfiber: yes |
| 23:59.45 | jayk2 | oh i thought maybe it was dialplan.xml |
| 23:59.56 | twisted[work] | dialplan.xml is for crisco |