| 00:00.07 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 00:00.59 | beek | pabelanger: This is *nix, so that's kind of the way it works. Once again, I prefer putting my apps into /usr/local or /opt to make it super-simple to back up. I can symlink to the usual spots as needed. |
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| 00:52.12 | lesterc | Can anyone recommend a ISDN card to be used under 64bit Ubuntu/Linux with asterisk 1.4? |
| 00:52.31 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:53.13 | Strom_C | what kind of ISDN? |
| 01:01.05 | *** join/#asterisk damjan_ (n=damjan@217.16.95.15) |
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| 01:18.05 | lesterc | Strom_C: how do I find out? :) |
| 01:18.23 | Strom_C | um |
| 01:18.24 | Strom_C | well |
| 01:18.29 | Strom_C | are you ordering BRI or PRI? |
| 01:18.47 | lesterc | I didn't order it. |
| 01:19.07 | lesterc | but I vaguely recall BRI (not knowing what it is) |
| 01:19.27 | Strom_C | I recommend you find out what kind of circuit you're dealing with before you proceed |
| 01:20.13 | lesterc | Care to share how will that affects my purchase? |
| 01:20.33 | Strom_C | because BRI and PRI use entirely different kinds of interface cards |
| 01:20.45 | Strom_C | so knowing is just a bit important |
| 01:20.49 | lesterc | and where do I find more background reading on this "ISDN thing"? |
| 01:21.04 | Strom_C | ~101 |
| 01:21.04 | jbot | it has been said that 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 01:21.05 | Strom_C | ~thebook |
| 01:21.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:21.35 | lesterc | Good bot. :) |
| 01:21.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:26.23 | xiando | speaking of cards, I have this PCI "modem" (yeah, analog modem thing) which apparently has sound card connectors. Is it possible to make that do something interesting with asterisk somehow? |
| 01:27.53 | [TK]D-Fender | xiando: No. |
| 01:28.41 | Qwell | xiando: is it a sound card? |
| 01:29.28 | Qwell | if it's an actual alsa supported sound card, you could probably use it for the console channel driver. basically though - no |
| 01:29.58 | Qwell | You aren't going to hook your phone up to it or anything though, if that's what you were thinking |
| 01:34.42 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 01:36.00 | xiando | I found the card and googled it and it's http://www.modem-help.co.uk/Askey/V1456VQH89B-Lucent-Win-Modem.html technology. It has two phone connectors on the back and a audio cable which looks like the audio cable which typically connects a CD-Rom device to a soundcard. |
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| 01:52.46 | *** mode/#asterisk [+o bkruse] by ChanServ |
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| 01:56.35 | [TK]D-Fender | xiando: Useless. Now forget about it and move on. |
| 01:57.09 | voxter | just stumbled upon a remote root exploit for trixbox that everyone to date is vulnerable to |
| 01:57.29 | voxter | it just so happens that fonality has already been notified, but its not likely that enough people are going to patch this |
| 01:57.32 | voxter | ...yikes. |
| 01:57.58 | Strom_C | well, that's their fault for running trixbox |
| 01:58.03 | voxter | indeed |
| 01:58.09 | xiando | voxter: lots2boxen4u, then. congratulations. have fun. remember to (ab)use torproject or similar when r00ting the trixbox boxen. |
| 01:58.41 | [TK]D-Fender | voxter: And .... who cares what happens to trixbox users anyways? |
| 01:58.49 | xiando | [TK]D-Fender: thanks for that advice. I'll forget about it after I make a video of me smashing it with a hammer (why just throw good hardware when you can videotape yourself smashing it?) |
| 01:59.05 | voxter | Its a vulnerability in the smarty PHP library for doing i8n |
| 01:59.11 | [TK]D-Fender | xiando: www.break.com. And if its good, link it on digg. |
| 01:59.28 | voxter | so im sure there are other projects exploitable that use smarty for languages. |
| 01:59.31 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
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| 02:05.36 | lesterc | Strom_C: Okay turns out that I have a ISDN modem here with 2 analog ports (Telstra NT1 PlusII) - so I should really be looking for a analog card. :) |
| 02:06.12 | Strom_C | lesterc: um |
| 02:06.14 | Strom_C | no |
| 02:06.27 | Strom_C | you don't want to convert your ISDN to analog before you connect it to your asterisk box |
| 02:06.30 | Strom_C | that's madness |
| 02:09.13 | lesterc | I know - but I just want to get it to work :) |
| 02:09.27 | Strom_C | sigh |
| 02:09.32 | Strom_C | don't make me bang my head into the desk... |
| 02:09.41 | lesterc | the ISDN is only used for security systems here anyway - which uses it as an analog line. |
| 02:10.07 | lesterc | should probably turn it an analog line - cheaper that way |
| 02:10.13 | Strom_C | NO |
| 02:10.17 | Strom_C | listen to me |
| 02:10.21 | Strom_C | you DO NOT want to do that |
| 02:10.29 | lesterc | yes I'm listening. :) |
| 02:11.58 | lesterc | may I ask why? |
| 02:13.10 | jaytee | man, I keep seeing tons of blocked events on all kinds of ports from IP addresses originating in China. |
| 02:13.29 | Nugget | My imitation of #asterisk: |
| 02:13.34 | Nugget | Hi everyone, I need help. I'm trying to run trixbox with realtime patches on an eMachines server I found in the storage closet. I've got 40 users with a mixture of grandstream phones and x-lite (unregistered). I'm using four clone x100p cards I bought off ebay and I compiled a pirated version of the g729 codec. Can you help me set up fax over sip? Please answer quickly because all our phones are down and my boss is getting pissed. Thanks!" |
| 02:13.51 | Strom_C | Nugget: ahaahahhaahahahahaaha |
| 02:13.54 | Strom_C | that's brilliant |
| 02:14.13 | jaytee | Nugget, it's a rerun but a great episode! :-) |
| 02:14.35 | xiando | "pirated" version of g729, you poor people in countries with patents. |
| 02:14.45 | Strom_C | lesterc: analog sounds like crap...the signaling is slower and more prone to error... |
| 02:15.00 | Nugget | if it weren't for software patents you'd be trying to build a phone system on an abacus. |
| 02:15.17 | Strom_C | xiando: please reach for the pompousness dial and turn it down towards "low" |
| 02:15.46 | jaytee | at least us poor people live in countries where someone's intellectual property isn't treated like state property like in some godless heathen communist country like China |
| 02:15.54 | Nugget | heh |
| 02:15.56 | lesterc | Nugget: people working in small shop face these kind of crap all day... |
| 02:16.15 | lesterc | Strom_C: got it. |
| 02:16.55 | lesterc | How crap are we talking here? :) |
| 02:17.11 | Strom_C | look |
| 02:17.14 | Strom_C | you have ISDN |
| 02:17.18 | Strom_C | don't waste your time with analog |
| 02:19.01 | lesterc | nod |
| 02:19.46 | lesterc | so what sort of h/w will I need? I was told that it's a passive ISDN (BRI?). |
| 02:20.44 | lesterc | phones uses voip in production - all I want to do is to provide a backup route when things go wrong (eg. emergency call etc) |
| 02:21.00 | Strom_C | get a digium b410p |
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| 02:27.30 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-70-106.sttlwa.fios.verizon.net) |
| 02:32.26 | philippel | how can I set a custom devstate through the manager? |
| 02:36.35 | [TK]D-Fender | philippel: http://www.voip-info.org/wiki/index.php?page_id=4398&tk=f5c568ff56f316305b91&comments_page=1 |
| 02:36.51 | [TK]D-Fender | philippel: for 1.4 1.6 latest beta supprts a CLI call for it natively. |
| 02:39.57 | voxter | anyone know off the top of their head how to set arbitrary variables on an IAX connection to pass variables between servers? |
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| 02:40.53 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 02:42.05 | [TK]D-Fender | voxter: I can see this in SIP, but not IAX |
| 02:42.36 | voxter | [TK]D-Fender: SIP allows you to set custom headers easily.. I was under the impression it could be done somehow in IAX as well, but i forget how. |
| 02:44.39 | voxter | Ah, shit. It's in 1.6 |
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| 02:51.15 | philippel | [TK]D-Fender I'm on 1.4 and there is only funcdevstate list available and trying change doesn't work, but if I treat it like setting a global variable it works fine |
| 02:51.53 | philippel | core set global DEVSTATE(Custom:DEVDND225) BUSY |
| 02:51.58 | philippel | that works on 1.4 |
| 02:53.51 | philippel | [TK]D-Fender I see it was dated April, looks like the change option did not get backported to russelb's 1.4 port |
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| 02:58.48 | [TK]D-Fender | philippel: good, you CAN read :) |
| 02:58.54 | [TK]D-Fender | philippel: indeed it was 3rd part |
| 02:58.58 | [TK]D-Fender | party* |
| 02:59.23 | [TK]D-Fender | philippel: So backport the app from that separate branch or wait around for 1.6 to hit a spot you're comfortable with |
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| 02:59.46 | philippel | no need, you can use set global variable to set it and it works fine |
| 03:00.20 | philippel | which actually makes it somewhat redundant:) |
| 03:01.30 | [TK]D-Fender | philippel: I guess if you want that applied at a given point in time. Keep in mind how that can become reset |
| 03:01.52 | [TK]D-Fender | philippel: Then again, you asked about a means of doing something, not the reason you were considering such means. |
| 03:02.15 | [TK]D-Fender | philippel: let alone getting any agreement that the approach was advisable. |
| 03:02.31 | [TK]D-Fender | philippel: So um yeah... hope that works out for you :) |
| 03:02.37 | philippel | [TK]D-Fender yes - I want it applied at the specifc time. Think of DND, if I hit the DND feature code on the phone, I want the state to change. NOw if I go to a web protal and hit DND through the manager, I want it applied at that point |
| 03:03.13 | philippel | I'm just trying to update the ARI with DND and some other feature codes now that I have added support all over for func_devstate in freepbx |
| 03:03.25 | [TK]D-Fender | event driven would make use of the CLI backport |
| 03:04.04 | philippel | maybe I'm confused? It looks to me that as soon as I set the value, through the manager using the CLI command the blf changes value? |
| 03:04.17 | philippel | the blf that subscribes to the hint that uses that state that is |
| 03:04.40 | philippel | so when you say event driven - am I missing something subtle? |
| 03:05.19 | [TK]D-Fender | philippel: If you want it called off a web-script, etc, then you'd need something AMI etc would be capable of doing to toggle it live. |
| 03:05.28 | [TK]D-Fender | philippel: So the backport would be a viable means. |
| 03:05.44 | [TK]D-Fender | philippel: There are more painful options, but... you don't want to do that I'm sure. |
| 03:05.51 | philippel | but I used the set global command through the AMI and it worked fine? |
| 03:06.49 | philippel | I just modified the Astersk CLI module today to run everything through the manager to address a security vulnerability that was there, so it makes it really easy to test anything through the AMI now:) |
| 03:08.11 | [TK]D-Fender | philippel: Ok, if that does it. |
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| 05:16.26 | j0 | anyone have a preference for wireless headsets? i'm looking for one that goes over the head |
| 05:21.36 | dominic1 | gn netcom |
| 05:21.55 | dominic1 | I am using gn9350 |
| 05:21.59 | dominic1 | I like it very much |
| 05:22.18 | dominic1 | [Jul 16 07:20:33] NOTICE[20260]: chan_iax2.c:5494 register_verify: Peer 'iaxmodem3' is not dynamic (from 127.0.0.1) |
| 05:22.24 | dominic1 | what does that mean?? |
| 05:23.32 | j0 | dominic1: thanks.. i didn't realize they were made by jabra.. cool |
| 05:24.19 | dominic1 | I always had headsets from gn netcom, after a long phase of testing others I always came back to gn |
| 05:24.26 | dominic1 | I think they are the best |
| 05:24.41 | j0 | did you ever use any plantronics? |
| 05:25.44 | dominic1 | I think I had one with cable, but I can't really remember |
| 05:26.07 | j0 | cool.. thanks very much.. i'll be buying a few gn9350's.. just saw that u can get a headset thing for em |
| 05:27.00 | j0 | dominic1: 1 more question :) where'd u buy yours? |
| 05:27.15 | dominic1 | I got mine from a german reseller called voipango |
| 05:29.44 | dominic1 | if you wan't to hang up calls etc.. you will need a cable between your headset and the phone |
| 05:30.10 | j0 | dominic1: i'm using polycom 501 and 601's.. i'm not sure if they support hangup, etc. though |
| 05:31.14 | dominic1 | started google: http://www.twacomm.com/catalog/model_14201-17.htm |
| 05:31.29 | dominic1 | think that should work.... |
| 05:31.50 | j0 | aw crap.. only on the higher models.. thanks though :) |
| 05:35.03 | dominic1 | no problem |
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| 06:12.50 | kamanashisroy | does asterisk update cdr in database on hangup ? I mean can I get the cdr from database in deadagi after call ? |
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| 06:41.25 | *** join/#asterisk Sam_N (n=Sam_N@62.24.111.38) |
| 06:42.09 | Sam_N | hi all |
| 06:42.15 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 06:42.28 | Sam_N | I need help with extensions |
| 06:44.48 | Sam_N | is it possible to do "exten => _X.,2,Set(${EXTEN}=${EXTEN:10}) " ? |
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| 06:49.26 | Sam_N | anyone there ? |
| 06:49.47 | kamanashisroy | Sam_N: see channelvariable.txt in asterisk doc |
| 06:54.04 | Sam_N | is it possible to change the value of ${EXTEN} and if so how ...an example will be appreciated |
| 06:55.25 | Sam_N | or even set the variable ${EXTEN} |
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| 06:57.52 | [netman] | Sam_N: why don't use Dial(....${EXTEN:10}....) ? |
| 06:58.15 | kamanashisroy | Sam_N: no you cannot write/set EXTEN .. |
| 06:58.36 | kamanashisroy | Sam_N: try do jump .. goto(${EXTEN}) ... |
| 06:59.02 | kamanashisroy | Sam_N: goto(${EXTEN:10}) ... please see the specification of goto |
| 06:59.23 | dominic1 | anybody with knowledge in hylafax here? |
| 07:00.10 | [netman] | dominic1: u could try also #hylafax |
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| 07:26.04 | gnorbert | Hi, How can I see in Asterisk, what kind of codec is used by a connection? |
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| 07:34.01 | JerJer | gnorbert: core show channel [channel idenifiter] |
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| 07:34.14 | gnorbert | JerJer: Thanks. |
| 07:34.35 | JerJer | something like ReadFormat and WriteFormat its gonna be in the bitmask format - so like 256 is G.729 |
| 07:34.38 | JerJer | 4 is ulaw |
| 07:34.59 | JerJer | 8 is alaw |
| 07:35.55 | JerJer | any diggs would be appreciated: http://tinyurl.com/624lrm |
| 07:36.05 | gnorbert | Thank you. |
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| 07:48.23 | Ast001 | [Jul 16 09:47:50] WARNING[16779]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 07:48.54 | Ast001 | now i can not connect analog with x100p card |
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| 07:51.17 | Ast001 | /etc/asterisk/zapata,conf http://pastebin.com/d35317dd9 |
| 07:51.44 | Ast001 | /etc/zaptel.conf http://pastebin.com/d2f1bc64a |
| 07:52.37 | Ast001 | ztcfg -vvvv http://pastebin.com/d592be506 |
| 07:52.42 | gramulhaozin | yaba daba doooooooooooo |
| 07:53.42 | Ast001 | lsmod | grep zaptel http://pastebin.com/ |
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| 08:35.04 | Sam_N | Can someone assist me with the following configuration ...asterisk receives a call 123456, strips 123,forwards 456 as the agi_extension to an agi script |
| 08:39.04 | Sam_N | anyone there ? |
| 08:40.58 | gnorbert | I am here, but I can't help you, sorry.:) |
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| 08:47.25 | dominic1 | will I have to add anything in asterisk to use did based routing in hylafax? |
| 08:47.42 | dominic1 | currently I am dialing IAX2/iaxmodem |
| 08:47.46 | tzafrir_laptop | a variation on: AGI(${EXTEN:3}) |
| 08:48.19 | tzafrir_laptop | err... completely wrong syntax. Ignore that |
| 08:49.23 | tzafrir_laptop | Goto(${EXTEN:3} ; does work, tohugh. |
| 08:54.22 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 08:54.43 | tzafrir_laptop | dominic1, a DID in chan_zap? |
| 08:55.44 | tzafrir_laptop | if a call comes in to DID 123456 and that channel is set to context 'incoming', then it will go to extension 123456 in context incoming |
| 08:56.11 | tzafrir_laptop | So put an explicit extension 123456 to handle the specific DID |
| 08:56.33 | tzafrir_laptop | And maybe a catch-all extension _X. to catch all other. |
| 08:56.37 | tzafrir_laptop | others, that is |
| 08:59.05 | tzafrir_laptop | Sam_N, something like: exten => 456,1,AGI(script_name.agi) |
| 09:01.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:04.39 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 09:13.10 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 09:15.07 | Sam_N | tzafrir, what if the 456 is a random number ? |
| 09:18.46 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:20.30 | *** join/#asterisk Gwayne (n=Gwayne@bb220-255-225-154.singnet.com.sg) |
| 09:21.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:21.34 | Gwayne | Maybe a silly question but what is the most stable version of asterisk 1.4 ? :) |
| 09:22.26 | *** join/#asterisk MoutaPT (n=m@193.136.44.104) |
| 09:23.56 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 09:25.31 | MoutaPT | hello, any one here experienced with sangoma cards runing on asterisk? |
| 09:25.54 | tzafrir_laptop | Sam_N, you make it sound as if http://xkcd.com/221/ :-) |
| 09:26.27 | mort_gib | MoutaPT: Yes... What's up?? |
| 09:27.09 | tzafrir_laptop | Sam_N, why do you want to "go to" an AGI extension? |
| 09:28.43 | MoutaPT | Problem with a MGW(asterisk) runing sangoma A 500, Call comes from PSTN-->MGW--->LegacyPBX--->real phone |
| 09:29.18 | MoutaPT | MGW is answering the call when it does Dial(woomera/...) to the legacy, and before the real phone answers the call |
| 09:29.19 | MoutaPT | :) |
| 09:29.20 | mort_gib | MoutaPT: -So where does * fit into that? |
| 09:29.25 | MoutaPT | * is the MGW |
| 09:29.32 | mort_gib | Ah, okay... |
| 09:29.59 | tzafrir_laptop | Sam_N, Please explain why you need that? |
| 09:30.18 | mort_gib | So you have one cable connected to your ISDN line (from Telco) and another connected to your "LegacyPBX" |
| 09:30.24 | MoutaPT | right |
| 09:30.45 | mort_gib | And you have spans configured correctly?? |
| 09:30.46 | MoutaPT | dial plan runing with digium TE4XX run smoothly |
| 09:31.07 | MoutaPT | i'm not on site, but as far as i have been told the config is ok |
| 09:31.21 | MoutaPT | all runing well besides the problem the call being answered |
| 09:31.29 | MoutaPT | by the MGW before it should be... |
| 09:31.30 | mort_gib | Can you dial phones in your LegacyPBX from * (softphone/IP phone) ? |
| 09:31.54 | mort_gib | -Before It should be?? please elaborate... |
| 09:32.29 | MoutaPT | this MGW has: exten=>_X.,1,Dial(Woomera/...) |
| 09:32.45 | MoutaPT | so the call comes from telco |
| 09:32.49 | MoutaPT | and reaches this context |
| 09:32.52 | MoutaPT | and dial starts |
| 09:32.53 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 09:33.21 | mort_gib | Yes, and?? |
| 09:33.24 | MoutaPT | just imediately dial starts and the call hasn't been answered in the phone from the legacy pbx, MGW answers it |
| 09:33.30 | MoutaPT | while dial still running |
| 09:33.44 | MoutaPT | and I get charged for a call that hasn't been answered yet |
| 09:33.47 | MoutaPT | got it? |
| 09:34.12 | MoutaPT | MGW should remain ringing for the telco |
| 09:34.22 | MoutaPT | and not answer... |
| 09:34.24 | mort_gib | Yeah, convince your client to dump their legacy system! |
| 09:34.32 | MoutaPT | impossible |
| 09:34.45 | MoutaPT | but that's a matter of sangoma |
| 09:34.54 | mort_gib | I don't really think you can avoid that... |
| 09:34.57 | MoutaPT | with digium TE... boards is runing fine |
| 09:35.14 | MoutaPT | so i believe some flag in the driver... |
| 09:35.15 | mort_gib | -So you are saying that Digium hardware acts differently?? |
| 09:35.20 | MoutaPT | yes |
| 09:35.34 | MoutaPT | on my office i have similar setup |
| 09:35.40 | MoutaPT | with E1 card |
| 09:35.45 | MoutaPT | and no problem |
| 09:35.52 | MoutaPT | this site is BRI card sangoma |
| 09:35.53 | mort_gib | Yes, very possibly, you will want to speak to Mr. Konrad Hammel (Sangoma Support) |
| 09:36.25 | MoutaPT | do you have his contact? or is he usually here? |
| 09:37.00 | mort_gib | -Eh, support@sangoma.com |
| 09:37.24 | mort_gib | konrad@sangoma.com |
| 09:37.46 | MoutaPT | technical team from my department i believe is already in contact i was just trying to go faster |
| 09:38.03 | MoutaPT | irc and chat usually helped me in the past |
| 09:38.17 | MoutaPT | so i decided to come back :) |
| 09:38.25 | MoutaPT | long time ago i didn't come here |
| 09:38.33 | MoutaPT | thks mort_gib |
| 09:39.00 | mort_gib | Your welcome |
| 09:39.33 | mort_gib | This kind of issue is like looking into the conf files and the documentation, time consuming and you will have to play around a bit |
| 09:39.45 | mort_gib | -Remember to use the LATEST wanpipe version!!! |
| 09:43.12 | jblack | hmm. Does anyone know if sip communicator works in mozilla? |
| 09:43.28 | jblack | I'd think the sandbox would prevent access to the microphone |
| 09:48.07 | *** join/#asterisk friendly12345 (n=friendly@ppp59-167-145-230.lns4.mel6.internode.on.net) |
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| 10:13.56 | tompaw | Morning. |
| 10:15.58 | jblack | yes. Hello. |
| 10:21.52 | Gwayne | Ahum evening :) |
| 10:32.58 | Takapa | is using global variables the only way of making sure a variable will survive a goto to another extension/context? |
| 10:33.32 | Takapa | i miss some sort of context inheritance (like channel inheritance) |
| 10:37.10 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
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| 10:39.52 | Takapa | (nm, logical flaw) |
| 10:41.24 | *** join/#asterisk rdsousa (n=chatzill@213.63.47.164) |
| 10:42.13 | rdsousa | hello everyone |
| 10:47.58 | Rico29 | http://debian.pastebin.com/m5ae0ee14 |
| 10:48.08 | Rico29 | can somebody explain this ? |
| 10:48.15 | Rico29 | please |
| 10:49.07 | tzafrir_laptop | I don't read French |
| 10:49.50 | Rico29 | there is juste 3-4 words of french... |
| 10:50.09 | Rico29 | I think it's easily understandable |
| 10:50.28 | Rico29 | (mmh is it correct english ?) |
| 10:50.32 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 10:50.39 | PodMan99a | can I use asterisk to provide my users with dialup internet? |
| 10:51.12 | *** join/#asterisk XnOSX (n=XnOSX@212.145.170.219) |
| 10:51.23 | rdsousa | does anyone have experience with Sangoma A500 BRI card? |
| 10:52.16 | Rico29 | tzafrir_laptop : http://debian.pastebin.com/m60dbff2b |
| 10:52.18 | Rico29 | ... |
| 10:53.52 | *** join/#asterisk Illarane (n=heifer@pdpc/supporter/student/Veratien) |
| 10:57.17 | Illarane | Probably not the best place to ask, but I figured you lot are more likely to know the answer than the bods in #defocus. :) I'm trying to write a bit of code that pulls out the correct flag for a phone number based on its IDD, though this is apparently impossible, as both the USA and Canada are +1, and loads of other codes also start with +1. :/ Is there some kind of logic to them, or a way to tell them apart without knowing adding eve |
| 10:58.52 | tompaw | Illarane: well, I'd suspect there is some mapping that distinguish US from Canda. But if there isn't - there are 3rd party services like Number Portability Lookup that get you the country name of requested phone number. |
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| 11:32.09 | test333 | hi, i've got a newbee question, is this the place to come for help? |
| 11:32.12 | *** join/#asterisk XnOSX (n=XnOSX@212.145.170.219) |
| 11:34.25 | gnorbert | Depends on the theme of the question, but I'm newbee too, so it's possible.:) |
| 11:36.03 | test333 | there is a problem a can't seem to resolve, have research and trail-and-errored for too long now |
| 11:36.31 | test333 | and i'm not enough of a programmer yet to make sense of asterisk sourcecode |
| 11:37.03 | test333 | what application, function, ... makes the following possible |
| 11:38.31 | test333 | dial somebody, talk to him/her and then redirect the dialed person to a different context but being able to hangup yourself |
| 11:38.38 | test333 | is that possible in asterisk |
| 11:39.04 | test333 | i tried all sorts of transfer but can't find enough info about it only |
| 11:39.52 | test333 | features.conf and all of those, my phone says transfer but that's where it stops |
| 11:40.00 | test333 | wrong extension... |
| 11:40.58 | test333 | there is probably not one solution, but a combination of steps...??? |
| 11:42.31 | tompaw | do you guys know any european PSTN terminator that lets you set your own CID? |
| 11:42.41 | tompaw | (during each call) |
| 11:43.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:43.13 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 11:56.27 | test333 | in summary: how to transfer a dialed person to an IVR after having spoken to them |
| 11:56.55 | test333 | or can you hangup on somebody but leave them connected to asterisk for the rest of the context? |
| 11:59.07 | deever | anyone with asterisk on ubuntu 8.04 here who brought sending faxes with hylafax to work? |
| 12:00.14 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
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| 12:07.31 | *** join/#asterisk twitchnln (i=42bbb3e2@gateway/web/ajax/mibbit.com/x-db51d0c0ece208c0) |
| 12:07.50 | twitchnln | morning |
| 12:08.00 | Segnale007 | hey guys |
| 12:08.02 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:08.04 | Segnale007 | I have a question |
| 12:08.28 | Segnale007 | does the digium pci analog cart work on sparc64 architetture ? |
| 12:09.08 | Segnale007 | *cards |
| 12:09.25 | tzafrir_laptop | Segnale007, Linux or Solaris? |
| 12:09.37 | Segnale007 | freebsd |
| 12:09.40 | twitchnln | I'm having a strange issue when handing calls off to from asterisk to televantage pbx via t1, the call connects, i hear the first 2 sec of the ivr and then * hangs up the channel, something very similar happens on outbound calls from the televantage system, anyone ever seen anything like that? |
| 12:09.44 | tzafrir_laptop | Linux: it should . MAybe it was even actually tested |
| 12:09.52 | Segnale007 | solaris ? |
| 12:10.01 | Segnale007 | I don't use linux .. |
| 12:10.31 | Segnale007 | should it work on solaris and sparc64 ? |
| 12:10.42 | tzafrir_laptop | I have no idea |
| 12:10.52 | Segnale007 | uhm ok .. |
| 12:10.58 | Segnale007 | Ill call digium then .. |
| 12:11.13 | tzafrir_laptop | ask on the silent #asterisk-bsd channel? |
| 12:11.28 | Segnale007 | anyway I am interested to make it work on freebsd and sparc64 |
| 12:11.30 | Segnale007 | oh |
| 12:11.31 | kamanashisroy | did you people synchronized manager script with dialplan ? |
| 12:11.45 | Segnale007 | I dind't know that exist and asterisk bsd channel |
| 12:11.50 | Segnale007 | ty |
| 12:11.52 | Segnale007 | Ill ask |
| 12:12.30 | tzafrir_laptop | anyway, what is the most recent version of chan_unicall? |
| 12:12.43 | russellb | if you find FreeBSD drivers, it might work, but will not be officially supported by Digium |
| 12:12.43 | tompaw | can I use a CIDR range in the 'host' field of sip.conf? |
| 12:12.52 | russellb | Segnale007: ^^ |
| 12:13.24 | *** join/#asterisk RobH (n=RobH@212.12.238.163) |
| 12:13.27 | [TK]D-Fender | tompaw: no |
| 12:13.56 | [TK]D-Fender | tompaw: an outbound call has to go to a single host/IP |
| 12:13.58 | tompaw | [TK]D-Fender: crap. Can I set the 'host' field multiple times regarding one peer? |
| 12:14.10 | tompaw | [TK]D-Fender: it's about incoming calls. |
| 12:14.19 | [TK]D-Fender | tompaw: for inbound you can use "dynamic" and then restrict based on a mask. |
| 12:14.48 | tompaw | [TK]D-Fender: I see, thanks. |
| 12:14.52 | Segnale007 | russelb |
| 12:14.52 | [TK]D-Fender | tompaw: Go read the WIKI for a list of fields. The ones you need will be fairly obvious |
| 12:14.56 | Segnale007 | I guess so ... |
| 12:15.00 | Segnale007 | Ill try then |
| 12:15.06 | Segnale007 | I have an old ultra 10 |
| 12:15.20 | Segnale007 | and I want to build up and fxo gw on it |
| 12:15.25 | tompaw | [TK]D-Fender: will do, thx. |
| 12:15.38 | [TK]D-Fender | twitchnln: Pastebin a failed call at verbose 10, and intense PRI debug enabled |
| 12:15.41 | Segnale007 | I can find zaptel drivers in freebsd ports for sparc64 |
| 12:15.58 | Segnale007 | then . I guess it should be work .. |
| 12:16.27 | test333 | can anybody point me to an article explaining a solution for my problem. I checked voip-info, asteriskguru, ... couldn't find one... |
| 12:16.39 | Segnale007 | I dont want to replace it for an x86 architetture .. |
| 12:16.56 | Segnale007 | I prefer my old sparc64 instead of a new x86 cpu |
| 12:17.11 | tzafrir_laptop | Segnale007, I don't know those specific drivers, but I figure it is worth a shot simply building the existing drivers for sparc64 |
| 12:17.21 | tzafrir_laptop | At worst they'll crash your system |
| 12:17.31 | Segnale007 | exactly |
| 12:17.43 | Segnale007 | I have to try and see what will happen |
| 12:17.48 | Segnale007 | that's all :) |
| 12:17.49 | [TK]D-Fender | And you'll have a card that you can't use in it. |
| 12:18.07 | Segnale007 | then Ill use an x86 cpu |
| 12:18.14 | Segnale007 | Ill buy an old one |
| 12:18.18 | Segnale007 | cheaper |
| 12:19.52 | tompaw | [TK]D-Fender: the thing is I was using sip.conf's content to define incoming DID calls as one peer with G729 forcing. As I changed host to dynamic/deny/permit - it's not recognized any more (falls into default context). |
| 12:20.42 | twitchnln | [TK]D-Fender: http://pastebin.ca/1073288 |
| 12:20.45 | [TK]D-Fender | tompaw: then you'd better look closer at what you've done. |
| 12:21.25 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 12:22.01 | tompaw | [TK]D-Fender: that's what I've done http://pastebin.com/m13b899e8 |
| 12:23.59 | tompaw | [TK]D-Fender: after http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels: "...then tries to find a [peer] section matching the caller's IP address..." |
| 12:24.37 | [TK]D-Fender | twitchnln: 1934 > start of a remote side disconnect. And please DON'T include 1600 lines of just at the start next time. |
| 12:25.46 | [TK]D-Fender | tompaw: Where's the call SIP debug in there? nd you should use "type=user" for this |
| 12:27.03 | tompaw | why? I don't want it to match basing on username (did doesn't register, just calls is) |
| 12:27.32 | tompaw | [TK]D-Fender: it was working fine when I had host = __ip__ instead of dynamic. that's why I asked about the CIDR range. |
| 12:27.39 | twitchnln | [TK]D-Fender: if the disco is coming from the remote side, any idea what would be causing it? as it happens in the midst of the ivr playing |
| 12:28.16 | [TK]D-Fender | twitchnln: No idea offhand. Clean up your PB, and keep it around. |
| 12:28.32 | [TK]D-Fender | twitchnln: I'm not an expert on this, but this is what you should have to be able to get help. |
| 12:28.49 | [TK]D-Fender | tompaw: Show me the call.... |
| 12:29.44 | tompaw | [TK]D-Fender: http://pastebin.com/m4ad9fff1 |
| 12:30.49 | tompaw | [TK]D-Fender: see line 59: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263) |
| 12:30.59 | tompaw | it's the default context :< |
| 12:31.29 | kamanashisroy | anyone here write perl ami script ? .. how to write test cases there ? |
| 12:32.04 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 12:32.07 | [TK]D-Fender | tompaw: Now set it to "user" |
| 12:32.13 | tompaw | ok |
| 12:33.01 | tompaw | [TK]D-Fender: still the same. |
| 12:34.10 | deever | anyone with asterisk on ubuntu 8.04 here who brought sending faxes with hylafax to work? |
| 12:36.05 | twitchnln | [TK]D-Fender: another thing i'm noticing is that after the call disconnects most of the time the b-channels restart |
| 12:37.14 | twitchnln | [TK]D-Fender: and a few times it has dropped the d-channel |
| 12:39.28 | [TK]D-Fender | tompaw: pastbin [general] and all related entries in sip.conf for this ITSP |
| 12:39.36 | tompaw | ok |
| 12:41.02 | tompaw | [TK]D-Fender: http://pastebin.com/m35b8fc81 |
| 12:41.47 | *** join/#asterisk domdom1986 (n=a@p5B3BE913.dip.t-dialin.net) |
| 12:41.48 | [TK]D-Fender | tompaw: Where is the non G729 entry? |
| 12:42.29 | domdom1986 | Hello guys! i have a question. who can help me with my easy Asterisk ... and outgoing calls |
| 12:42.32 | domdom1986 | ? |
| 12:43.41 | [TK]D-Fender | domdom1986: PASTEBIN is your friend. Show us your problem and maybe we can help |
| 12:43.43 | [TK]D-Fender | ~pb |
| 12:43.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:43.45 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 12:43.56 | tompaw | [TK]D-Fender: as you see, [general] doesn't say anything about codecs, so I guess this gsm|ulaw|alaw|h263 is a default set when nothing's defined. |
| 12:44.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:44.27 | [TK]D-Fender | tomtI just asked where your other entries for this ITSP were... |
| 12:44.36 | domdom1986 | Her the Question: I have a phonesystem ... to go out you dial a zero ... the asterisk is connected to the phonesystem ... i can dial numbers on the telephonesystem but not in the city |
| 12:44.43 | tompaw | [TK]D-Fender: what other entries? that's all. |
| 12:45.00 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 12:45.51 | [TK]D-Fender | domdom1986: pastebin the complete CLI output of your call attempt at verbose 10, and include your dialplan. |
| 12:46.15 | Yourname`` | Hi. Quick question, I have phones registered to Asterisk. A few of them are Polycom IP 330s, and then most of them are Aastra 9133is. Through the IVR when someone enters the extension, if it is Aastra, it rings. If it's the Polycom, it doesn't ring and sometimes it does ring. What kind of a problem is this and what can I do to solve? |
| 12:46.24 | [TK]D-Fender | tompaw: And if you comment out the allow/deny? |
| 12:46.47 | tompaw | [TK]D-Fender: before I try that - how the hell is it gonna recognize this call then? |
| 12:47.04 | kamanashisroy | Yourname``: may be you ring for longer time ? |
| 12:47.41 | Yourname`` | kamanashisroy: Sorry? |
| 12:48.26 | domdom1986 | Can somebody help me with my problem with my outgoing calls? i have created some patterns but i dont work! |
| 12:48.32 | kamanashisroy | Yourname``: what happens when it does not ring ? |
| 12:48.57 | kamanashisroy | Yourname``: does it succeed making outgoing call ? |
| 12:49.25 | Yourname`` | kamanashisroy: This is the dial string: exten => 10,1,Dial(SIP/10,20). Through the IVR if someone dials ext 10, it has dead air for 20 seconds and then says agent unavailable |
| 12:49.29 | [TK]D-Fender | tompaw: we will see if your entry is good at ALL. dynamic will allow ALL that can otherwise match your cal. |
| 12:49.30 | Yourname`` | And then goes to the next priority |
| 12:49.44 | [TK]D-Fender | domdom1986: and I asked you to show me your DIALPLAN. |
| 12:49.50 | tompaw | [TK]D-Fender: trying it now. |
| 12:50.33 | [TK]D-Fender | Yourname``: And the reason you haven't put up a pastebin of your entire dialplan and SIP debug CLI output of your cal is...? |
| 12:50.43 | kamanashisroy | Yourname``: if not unavailable then not unavailable .. if you think it were available then try increasing dial timeout .. or try increasing resgistration timeout .. (not sure about that) |
| 12:51.24 | tompaw | [TK]D-Fender: not working. it only recognizes this peer when there's an explicit ip address :-( |
| 12:51.34 | kamanashisroy | Yourname``: [TK]D-Fender is right |
| 12:51.58 | kamanashisroy | domdom1986: are you hiring ? |
| 12:52.18 | Yourname`` | lol [TK]D-Fender cuz you didn't ask for it yet. |
| 12:52.21 | [TK]D-Fender | tompaw: if all you allow is G729, why do you need to restrict to an IP range? |
| 12:52.38 | kamanashisroy | Yourname``: but you asked for help !! |
| 12:52.42 | [TK]D-Fender | Yourname``: Don't ask for an answer to a problem without SHOWING IT TO US |
| 12:53.13 | tompaw | [TK]D-Fender: I just want to control what falls into which basket, if you know what I mean. |
| 12:53.49 | kamanashisroy | [TK]D-Fender: do you like my answer for domdom1986 ? |
| 12:54.41 | [TK]D-Fender | kamanashisroy: yeah, ok, fine, sure.... |
| 12:55.35 | tompaw | [TK]D-Fender: it has to be 'peer' and explicit ip in 'host' field. I'll try multiple host fields, but I think it will cause a parse error. |
| 12:55.40 | Yourname`` | [TK]D-Fender: http://pastebin.ca/1073319 |
| 12:56.15 | domdom1986 | Hello my name is Dominik, |
| 12:56.16 | domdom1986 | i have some problems with my easy asterisk system. My asterisk is behind a telephone system. To dial out of the asterisk (outbound routing) you must dial a 0. To dial out of the telephone system you must also dial a zero. |
| 12:56.16 | domdom1986 | I think you must dial 00 to dial out... but it dont go. |
| 12:56.16 | domdom1986 | When i dial 0 i can communicate with the telephone system and the MSNs on the telephone system. |
| 12:56.16 | domdom1986 | I have created a new outbound routing with the pattern XXX this is for the internal telephone system communikation. Now i have created a pattern like [0]X for all other thinks in the wide world. |
| 12:56.18 | domdom1986 | But when i dial 00 0203 XXXXX it dont works. |
| 12:56.20 | domdom1986 | greetz |
| 12:56.22 | domdom1986 | Dominik |
| 12:56.30 | domdom1986 | http://pastebin.com/m7da6849d |
| 12:56.33 | *** join/#asterisk Peri (n=redanti@surf99.net.rss.rogers.com) |
| 12:56.33 | tompaw | or not, only the latest ip is saved to a peer :( |
| 12:56.54 | kamanashisroy | Yourname``: I do not see any queue or agent ! |
| 12:57.06 | [TK]D-Fender | domdom1986: I asked you to pastebin your DIALPLAN, not spam this channel with junk |
| 12:57.21 | [TK]D-Fender | domdom1986: do not do that again. |
| 12:57.37 | [TK]D-Fender | Yourname``: and the other half? |
| 12:58.03 | tzafrir_laptop | ~not work |
| 12:58.04 | jbot | Look buddy, doesn't work is an ambiguous statement. Does it sit on the couch all day? Does it want more money? Does it waste its time on IRC all day long? Please be specific! Define 'it' and what it isn't doing. |
| 12:58.11 | *** join/#asterisk TraydenT (n=traydent@212.73.216.2) |
| 12:58.19 | TraydenT | Hello |
| 12:58.46 | *** join/#asterisk CVirus (n=Satan@196.218.41.31) |
| 12:58.47 | tzafrir_laptop | hmm... not good enough |
| 12:58.50 | Yourname`` | [TK]D-Fender: of what? |
| 12:59.09 | kamanashisroy | Yourname``: did you try increasing dial timeout ? |
| 12:59.12 | [TK]D-Fender | Yourname``: And the reason you haven't put up a pastebin of your entire dialplan and SIP debug CLI output of your cal is...? <----------- |
| 12:59.20 | [TK]D-Fender | kamanashisroy: thats part of his problem. |
| 12:59.23 | Yourname`` | [TK]D-Fender: Ohh.. yeah still working on it |
| 12:59.31 | [TK]D-Fender | kamanashisroy: his dead air is IVR timeout. |
| 13:00.12 | kamanashisroy | domdom1986: did you try to write any dialplan ? |
| 13:01.16 | yangvnc | How do I fix error : |
| 13:01.19 | yangvnc | Registration from '<sip:6002@123.123.123.123;user=phone>' failed for '84.84.84.84' - Not a local domain |
| 13:01.20 | domdom1986 | no not yet |
| 13:01.24 | tzafrir_laptop | domdom1986, you use chan_zap, right? |
| 13:01.40 | tzafrir_laptop | what is the output of 'zap show channels'? |
| 13:02.00 | *** join/#asterisk ToTo (n=ToTo@209.8.41.108) |
| 13:02.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:02.25 | [TK]D-Fender | tzanger: He's got dialplan errors and has no clue about extensions.conf |
| 13:02.33 | domdom1986 | incoming calls are ok.. one mom for zap |
| 13:02.43 | [TK]D-Fender | domdom1986: GUI's are not supported in this channel. |
| 13:02.46 | xiando | yangvnc: add domain=<ip>, which likely isn't 84.84.84.84 |
| 13:02.47 | *** join/#asterisk rshortt (n=rob@freevo/developer/rshortt) |
| 13:03.09 | yangvnc | xiando: under users.conf account ? |
| 13:03.12 | [TK]D-Fender | tompaw: Why would calls from them be coming from a range of IP's? |
| 13:03.19 | Yourname`` | damn it's so hard to get CLI when the CLI is filled with other stuff |
| 13:03.22 | Yourname`` | onesec |
| 13:03.27 | xiando | yangvnc: under [general] in sip.conf |
| 13:03.29 | tompaw | [TK]D-Fender: http://www.didww.com/support/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=35&nav=0,11 |
| 13:03.34 | tompaw | [TK]D-Fender: probably some load balancing stuff. |
| 13:03.38 | TraydenT | Is it possible to have suppor tfor asterisk 1.6 there ? |
| 13:04.32 | domdom1986 | zap show channels |
| 13:04.33 | domdom1986 | <PROTECTED> |
| 13:04.33 | domdom1986 | <PROTECTED> |
| 13:04.33 | domdom1986 | <PROTECTED> |
| 13:04.33 | domdom1986 | <PROTECTED> |
| 13:04.33 | domdom1986 | <PROTECTED> |
| 13:04.35 | domdom1986 | <PROTECTED> |
| 13:04.46 | CVirus | Why am I getting so many "localhost kernel: FXO PCI Master abort" in my /var/log/messges ? |
| 13:04.47 | [TK]D-Fender | tompaw: And all their other guides show them telling you to add 1 peer for each. Guess thats what you're going to have to do. |
| 13:04.56 | [TK]D-Fender | domdom1986: STOP SPAMMING THIS CHANNEL. |
| 13:05.01 | dominic1 | problem with hylafax : textfmt: No font metric information found for "Courier-Bold" |
| 13:05.08 | dominic1 | after compiling it manually under debian |
| 13:05.08 | domdom1986 | i dont spamming i post something |
| 13:05.14 | dominic1 | can anybody help me with that? |
| 13:05.15 | domdom1986 | ^^ |
| 13:05.16 | [TK]D-Fender | domdom1986: 3 lines or more belongs in a pastebin. |
| 13:05.24 | tompaw | [TK]D-Fender: shame :) I think Asterisk should have support for CIDR ranges regarding incoming calls. |
| 13:05.28 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 13:05.32 | tompaw | [TK]D-Fender: as always, thanks for your help. |
| 13:05.48 | domdom1986 | http://pastebin.com/d1327a555 |
| 13:05.48 | [TK]D-Fender | domdom1986: pasting 6 lines in here like that is spamming. You will not be warned again. |
| 13:05.56 | yangvnc | xiando: thanks ! |
| 13:06.19 | domdom1986 | ok |
| 13:06.37 | Yourname`` | [TK]D-Fender: http://pastebin.ca/1073326 I was able to get this far |
| 13:06.40 | [TK]D-Fender | domdom1986: And until you can show us what you did in the dialplan that you believe should work yet doesn't, we cannot help you. |
| 13:06.58 | domdom1986 | ok 1 min |
| 13:07.34 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 13:07.36 | teknoprep | hey all |
| 13:07.42 | teknoprep | i ahve a question.. about transfering calls |
| 13:07.47 | domdom1986 | i have no dialplan |
| 13:08.10 | Yourname`` | lolol domdom1986 |
| 13:08.22 | teknoprep | i use the transfer softkey on the phone...warm transfer to another ext... ext picks up.. i say i have a call from so-and-so,,,, i hit transfer again and they get the call |
| 13:08.36 | domdom1986 | or you mean something .. i dont understand |
| 13:08.47 | Yourname`` | extensions.conf |
| 13:08.48 | teknoprep | now my question is... how would i get TRANSFER to play to the recievinv user when i hit finish my side of the warm transfer |
| 13:08.52 | [TK]D-Fender | Yourname``: -- Executing [s@ivrmenu:7] BackGround("SIP/64.74.188.22-b77084a8", "amsatnew") in new stack <-- your dialplan does not match what your CLI output shows. They are apples & oranges. |
| 13:08.56 | domdom1986 | i have a trunk configuration and a outbound routing |
| 13:09.03 | [TK]D-Fender | Yourname``: exten=>s,n(menu),Background(newivr) |
| 13:09.17 | teknoprep | [TK]D-Fender, whats up fender |
| 13:09.23 | russellb | teknoprep: i don't think there is a way to do that. |
| 13:09.28 | Yourname`` | [TK]D-Fender: Sorry that's me changing values so it's not googleable, lol |
| 13:09.33 | [TK]D-Fender | domdom1986: GUI's are not supported in this channel. Please read the channel topic and go to the appropriate support channel. |
| 13:09.39 | Yourname`` | amsatnew is the one |
| 13:09.40 | teknoprep | russellb, even if ALL of my phones are canreinvite=no |
| 13:09.42 | russellb | of course, between you saying "i'm about to transfer you", and then hearing music on hold ... |
| 13:09.45 | Yourname`` | And I messed up |
| 13:10.02 | TraydenT | I get an error with res_config_ldap on asterisk-1.6beta9 : [Jul 16 14:35:49] ERROR[5336]: res_config_ldap.c:1292 update_ldap: Couldn't modify dn:uid=chartonflo,ou=etude,ou=People,o=eisti,c=fr because Invalid syntax -- Registered SIP 'chartonflo' at 192.168.11.224 port 5060 expires 3600 |
| 13:10.29 | TraydenT | Asterisk tries to set an Atrribute with an empty value |
| 13:10.34 | teknoprep | russellb, pretty much there is NO indication of the transfer being finalized |
| 13:11.17 | domdom1986 | where i can find the config file of dialplan=? |
| 13:11.23 | teknoprep | russellb, so when our recep transfers a call to me... i get this... call comes in from recep... she says, hey i have a call for you... i say ok pass it on.... she hits transfer... i have NO INDICATION what-so-ever that i got this transfer... i just have to guess that it was completed... no beep no nothing |
| 13:11.57 | russellb | Ah, I see ... |
| 13:11.59 | [TK]D-Fender | Yourname``: Now for the third time I saked for SIP DEBUG <- |
| 13:12.13 | russellb | teknoprep: adding a beep there would be trivial i think ... |
| 13:12.22 | teknoprep | russellb, trivial ? |
| 13:12.25 | Katty | morning |
| 13:12.26 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 13:12.28 | russellb | sorry, thinking out loud |
| 13:12.30 | Yourname`` | [TK]D-Fender: Even though there's going to be a mixture of other things, is that ok? |
| 13:12.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.34 | russellb | it's not possible without modifying the source |
| 13:12.37 | teknoprep | russellb, why trivial... that would be GREAT.. BEST thing i have ever heard of |
| 13:12.40 | teknoprep | russellb, oh |
| 13:13.02 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-88-113.vif.net) |
| 13:13.06 | russellb | actually, now that i think about it, it would not be easy at all |
| 13:13.11 | [TK]D-Fender | Yourname``: For the 20 sec timeframe this should take and the amount of time wasted NOT giving what was requested I'm sure you could clean it up. |
| 13:13.12 | teknoprep | russellb, lol |
| 13:13.15 | russellb | :) |
| 13:13.17 | russellb | sorry ... |
| 13:13.31 | teknoprep | russellb, hey np... maby put it on the todo list ? i mean its not that bad of an idea |
| 13:13.51 | file | it's already been done and will be in 1.6.1 |
| 13:13.55 | teknoprep | russellb, its definetly somethign that would be worth having.. i listen to how some of these transfers get done sometimes in the office... its bad becuase ppl don't know when it finishes |
| 13:14.00 | russellb | file: srsly? |
| 13:14.02 | caio1982 | teknoprep: setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep doesn't help? |
| 13:14.02 | teknoprep | file, are you talking to me ? |
| 13:14.10 | Katty | file talks to himself. |
| 13:14.13 | file | ATTENDED_TRANSFER_COMPLETE_SOUND |
| 13:14.13 | Katty | i have seen it. |
| 13:14.18 | russellb | file: ha! |
| 13:14.24 | teknoprep | file, wooooooooohoooooooooooooo |
| 13:14.27 | Yourname`` | [TK]D-Fender: lol if I knew what is what, it might be easier.. but let me try sir, one sec |
| 13:14.28 | file | but like I said... |
| 13:14.28 | domdom1986 | ..hello? |
| 13:14.33 | file | it will be in 1.6.1 and above |
| 13:14.33 | teknoprep | file, russellb too bad i don't have a 1.6 installation |
| 13:14.36 | Katty | domdom1986: herro. |
| 13:14.43 | teknoprep | file, russellb, any rough ETA ? |
| 13:15.05 | russellb | a few months, at least |
| 13:15.09 | [TK]D-Fender | domdom1986: Go read the BOOK for a while. |
| 13:15.11 | [TK]D-Fender | ~book |
| 13:15.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:15.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 13:15.33 | tompaw | PDF books don't server well as pot holders :-( |
| 13:15.41 | russellb | file: oh, only for SIP transfers |
| 13:15.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:15.51 | teknoprep | 1.6 is going to be great.. with the SIP over TCP |
| 13:15.59 | file | easy to add for others |
| 13:16.01 | [TK]D-Fender | tomtoSure they do, especially when they "coaster" |
| 13:17.33 | [TK]D-Fender | domdom1986: Pay special attention to Chapter 5 |
| 13:17.48 | domdom1986 | ok thx |
| 13:20.09 | *** join/#asterisk freckle (n=jon@195.74.96.118) |
| 13:20.13 | Yourname`` | [TK]D-Fender: Still working on getting the debug.. one min, sorry.. (DID trouble now) |
| 13:20.47 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 13:21.10 | xiando | yeah SIP over TCP is great, you could in theory push it over Tor -- except that Tor exit nodes must be assumed hostile and eavesdropping, which makes the whole advantage of TCP irrellevant until Asterisk gets the way more important ZRTP/SRTP support. |
| 13:21.11 | tompaw | thinks that Asterisk's CLI would me much nicer if it had something like the 'script' shell command. |
| 13:22.07 | tzafrir_laptop | tompaw, look for astcli |
| 13:22.23 | russellb | or tee ... |
| 13:22.25 | [TK]D-Fender | xiando: Nope. |
| 13:22.37 | russellb | hands [TK]D-Fender an explosive tomato and runs |
| 13:22.49 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-121-0003.broadweave.net) |
| 13:22.51 | [TK]D-Fender | xiando: SIP is only your call setup. the voice itself is still UDP <_ |
| 13:23.19 | TraydenT | I get an error with res_config_ldap on asterisk-1.6beta9 : [Jul 16 14:35:49] ERROR[5336]: res_config_ldap.c:1292 update_ldap: Couldn't modify dn:uid=chartonflo,ou=etude,ou=People,o=eisti,c=fr because Invalid syntax -- Registered SIP 'chartonflo' at 192.168.11.224 port 5060 expires 3600 |
| 13:23.25 | tzafrir_laptop | tompaw, OTOH, that "script" may exist in the form of /var/log/asterisk/ |
| 13:23.41 | [TK]D-Fender | pulls the pin on the explosive tomato calculates russellb's exit point and throws. |
| 13:24.14 | tompaw | tzafrir_laptop: it can? to capture EVERYTHING that appears in the CLI? |
| 13:24.23 | tzafrir_laptop | tompaw, most things |
| 13:24.58 | *** join/#asterisk jg_uk (n=jmkg@62.3.216.18) |
| 13:25.56 | tompaw | tzafrir_laptop: how? ;) |
| 13:26.41 | jg_uk | has anyone had success compiling zaptel on linux 2.6.26? I see there is a bug about it |
| 13:26.59 | jg_uk | don't know if there are svn updates to address it yet |
| 13:27.16 | tzafrir_laptop | tompaw, by editing logger.conf, mostly |
| 13:27.25 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 13:27.26 | tzafrir_laptop | and running later 'logger reload" |
| 13:29.38 | tompaw | tzafrir_laptop: thx for the tip |
| 13:30.46 | rshortt | hi all, I have an idea for chan_mobile (or a new module). Use bluetooth to detect the presence of your mobile phone, if it's NOT there forward the call to it, otherwise let it go to your regular extension... |
| 13:30.49 | [TK]D-Fender | jg_uk: When running bleeding-edge tech, don't forget to insert the blad horizontally through the chest so you don't get stuck on the ribs. Then give it a good twist :) |
| 13:31.36 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:31.42 | [TK]D-Fender | rshortt: Sounds like something you could cron up completely separate from * |
| 13:31.55 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 13:31.58 | [TK]D-Fender | rshortt: On a 10s interval or something |
| 13:32.20 | rwaite | is there a way to ensure that something happens 'after' a Dial command? i mean one that successfully connects to the dialed extension? |
| 13:32.25 | rshortt | [TK]D-Fender: yeah possibly, or hack up a client app on peson's PC/laptop using bluetooth, send some presence info to * |
| 13:32.52 | jg_uk | [tk]D-Fender: not sure I would consider current stable to be equivalent to "bleeding edge" |
| 13:33.15 | [TK]D-Fender | rwaite: "g" dial option + "h" standard extension. |
| 13:33.48 | [TK]D-Fender | jg_uk: Just think about the "Debian way" and cringe then :) |
| 13:34.04 | jg_uk | I am using Debian... ;-) |
| 13:34.12 | tompaw | Phew. Freebsd 7 x86_64 is the real challenge. And keeping it legit. |
| 13:34.27 | tompaw | (re: g729 codecs) |
| 13:34.51 | tzafrir_laptop | jg_uk, hang on, giving it a shot... |
| 13:34.57 | Yourname`` | [TK]D-Fender: Still gettn the DID problem figured out |
| 13:35.25 | xiando | tompaw: if you're in EU then it's probably legit |
| 13:35.41 | [TK]D-Fender | Yourname``: Just come back when you actually have something to show :) |
| 13:35.54 | rwaite | this is ridiculously complicated :/ |
| 13:36.03 | tompaw | xiando: I am in EU. is it? |
| 13:36.11 | rwaite | i'm about to just say screw transfers all together |
| 13:36.31 | tompaw | xiando: I mean those: http://asterisk.hosting.lv/ codecs. They are the only one that work with my BSD. |
| 13:36.49 | [TK]D-Fender | tompaw: I think that "fair use" would allow to you use the "illegal" codecs out there if you just buy the licenses from Digium. Shouldn't matter that you aren't using their CODE. Just a thought. |
| 13:37.02 | tompaw | [TK]D-Fender: actually, that's what I did :D |
| 13:37.06 | Yourname`` | [TK]D-Fender: :) thanks |
| 13:37.30 | tompaw | [TK]D-Fender: I was looking for a way to buy the permission directly from the patent holder. But I don't think it's possible in EU. |
| 13:37.31 | [TK]D-Fender | tompaw: Fair is fair, and I would be very upset to hear if anyone has a real problem with this concept. |
| 13:37.56 | tompaw | [TK]D-Fender: I think what I'm paying for is the algorithm anyway, not the code. |
| 13:39.01 | [TK]D-Fender | tompaw: Well if the patent for the "illegal" one belongs to VoiceAge, and the same for Digium's I would believ them to be interchangeable. |
| 13:40.26 | tompaw | [TK]D-Fender: that's it. The algo is the same, isn't it? |
| 13:41.14 | [TK]D-Fender | tompaw: I think largely the same. Don't know the finer points, but this is very lightly "grey" at worst. How can you seriously fault someone for it? |
| 13:42.42 | tompaw | I'd love to hear somebody from Digium expressing their opinion about this case. |
| 13:43.26 | [TK]D-Fender | tompaw: Ask to speak to their legal dept... that should motivate them to invest in writing native code for it. Would be far cheaper :) |
| 13:43.29 | deever | anyone with asterisk on ubuntu 8.04 here who brought sending faxes with hylafax to work? |
| 13:43.48 | UnixDog | deever there are howto-s out there |
| 13:43.56 | [TK]D-Fender | deever: "brought"? Hylafax works, Ubuntu works, and Asterisk works. Whats your issue? |
| 13:44.00 | UnixDog | that explain how to setup and configure it |
| 13:44.14 | jg_uk | [TK]D-Fender: Spotted the problem yet? |
| 13:44.32 | *** part/#asterisk rshortt (n=rob@freevo/developer/rshortt) |
| 13:44.37 | deever | [TK]D-Fender: that the faxes don't get fed to asterisk |
| 13:44.53 | [TK]D-Fender | jg_uk: lol, no. I run CentOS (RHEL) and am a healthy number of steps back. |
| 13:45.07 | [TK]D-Fender | deever: And what have you shown us for your problem? |
| 13:45.09 | tompaw | [TK]D-Fender: good idea. I wonder if there are any differences in performance between those codecs (Digium vs .lv) |
| 13:45.24 | deever | althogh setup after an asterisk book |
| 13:45.27 | [TK]D-Fender | deever: PASTEBIN is your friend. |
| 13:46.01 | [TK]D-Fender | deever: And your description is unfortunately of no use so far. |
| 13:49.34 | UnixDog | look on voip-info.org |
| 13:49.43 | UnixDog | and voipcoop.org |
| 13:50.05 | UnixDog | and google asterisk+iaxmodem+hylafax |
| 13:51.01 | UnixDog | there are loads of how to setup and use hylafax with asterisk |
| 13:53.13 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:53.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.49 | beek | Hello UnixDog |
| 13:56.19 | UnixDog | Morning |
| 13:56.51 | UnixDog | well getting 1.6 to build and compile in the bsd ports tree is being a pain |
| 13:56.56 | beek | UnixDog: on the voip-users-conference you mentioned that you were starting to play with the Alix boards and *... were you satisfied with the results? |
| 13:57.08 | UnixDog | but I think I almost have it |
| 13:57.27 | UnixDog | yes I like the alix boards.. they work great |
| 13:57.37 | UnixDog | and they are simple to setup and use |
| 13:57.48 | beek | I think I'm going to take the plunge and order one to play with. |
| 13:58.12 | beek | Which of the boards to you find the most flexible? |
| 13:58.23 | UnixDog | I have a 2c2 and a 3c2 |
| 13:58.30 | UnixDog | and I like them both |
| 13:58.52 | UnixDog | I got full setups board+case+power brick |
| 13:59.07 | beek | That's what I'm going to do. |
| 13:59.07 | dominic1 | why do I always get this message in my log: [Jul 16 15:58:39] NOTICE[20257]: chan_iax2.c:5494 register_verify: Peer 'iaxmodem3' is not dynamic (from 127.0.0.1) |
| 14:00.42 | [TK]D-Fender | dominic1: Because it is registering, and you set a host to something other than "dynamic" |
| 14:00.51 | [TK]D-Fender | dominic1: Exactly like it says |
| 14:01.22 | dominic1 | is it wrong to set a host? |
| 14:01.34 | [TK]D-Fender | dominic1: if you have a fixed host, thats saying its shuoldn't be registering. Remember the the point of registering is so * knows where to send calls. It will always know if you tell it where. |
| 14:01.41 | [TK]D-Fender | dominic1: Yes, it is. |
| 14:02.05 | beek | UnixDog: thanks! |
| 14:02.07 | dominic1 | ah okay |
| 14:02.10 | UnixDog | beek np |
| 14:02.11 | dominic1 | I understand |
| 14:02.12 | [TK]D-Fender | dominic1: I would of course advise you to use a permit/deny mask to ensure that it is only local of course. |
| 14:02.22 | dominic1 | so I wil set it to dynamic and use permit |
| 14:02.28 | dominic1 | thank you very much |
| 14:02.37 | [TK]D-Fender | dominic1: np |
| 14:03.00 | dominic1 | should I use different modems for sending and receiving with iaxmodem? |
| 14:03.21 | UnixDog | no |
| 14:03.34 | UnixDog | you have to set iaxmodem up like a extension |
| 14:04.07 | UnixDog | you need to go read the howto's and fallow them work for word |
| 14:04.43 | dominic1 | in my extensions.ael I wrote the dial statement like that Dial(IAX2/iaxmodem1); Dial(IAX2/iaxmodem2) and so on. Should I check with chanisavail first? |
| 14:05.13 | tzanger | hmm |
| 14:05.34 | [TK]D-Fender | dominic1: I strongly suspect that IAXmodem will refuse extra calls to it. |
| 14:05.50 | [TK]D-Fender | dominic1: So it should be safe to call them back to back |
| 14:06.01 | [TK]D-Fender | dominic1: Go right ahead and test |
| 14:06.17 | dominic1 | :-D thank you fender, I will try that! |
| 14:08.18 | Katty | wibbles. |
| 14:08.21 | Katty | wobbles. |
| 14:08.44 | UnixDog | catches Katty so she dont fall down |
| 14:09.06 | Katty | kthx. |
| 14:09.19 | Katty | this day is so horrible, i want to start drinking now. |
| 14:09.30 | Katty | for the past week i've had NOTHING to do. maybe more like past 3 weeks. |
| 14:09.47 | Katty | except for the occasional My Outlook Wont' Work |
| 14:10.05 | UnixDog | well thats like me .. I tore my left knee and am stuck at home .. DR wont release me to go back to work |
| 14:10.14 | UnixDog | and its driving me insane |
| 14:10.18 | UnixDog | cabin fever |
| 14:10.21 | Katty | yikes. |
| 14:10.23 | Katty | poor knee!!! |
| 14:10.24 | UnixDog | 2 weeks now |
| 14:10.36 | Katty | i've been resorting to playing wow at work just to pass the time. |
| 14:11.09 | UnixDog | wow man that is slow |
| 14:11.16 | Katty | you have no idea. |
| 14:11.32 | Katty | like today, the only thing i have to do so far is backups (which is nearly done) and a quote for a company. |
| 14:11.50 | UnixDog | go refile all the filing cabnets according to report date and time and lastname of client then cross ref them with the issue |
| 14:11.53 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:12.00 | Katty | ^_- |
| 14:12.03 | Katty | we don't have file cabinents. |
| 14:12.04 | UnixDog | lol |
| 14:12.11 | Katty | i have pdfs. |
| 14:12.12 | UnixDog | ok the data bases then |
| 14:12.13 | Katty | and things. |
| 14:12.15 | UnixDog | ok |
| 14:13.18 | UnixDog | and open slacker.com on your desktop and stream music and dance around |
| 14:13.18 | lmadsen | I've worked from home for the past 4 years :) |
| 14:13.25 | UnixDog | get the whole office to join in |
| 14:13.32 | UnixDog | make it a office party |
| 14:13.33 | Katty | oh! but last weekend i was productive. we moved everything out of the basement and moved the computers, tv, etc down there. |
| 14:13.43 | UnixDog | ahh ok |
| 14:14.06 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:14.09 | Katty | i will never underestimate the pain of desk with hutch and stairs, ever, again. |
| 14:14.59 | UnixDog | lol |
| 14:15.01 | UnixDog | true |
| 14:15.09 | UnixDog | stairs are a pain |
| 14:15.15 | UnixDog | thats how I ripped my knee |
| 14:15.21 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 14:15.27 | Mike8861 | hello all\ |
| 14:15.28 | Katty | )= |
| 14:15.37 | Mike8861 | its raining day today |
| 14:15.39 | UnixDog | I was moving a loveseat slepper unit from the basement to the first floor |
| 14:15.50 | UnixDog | sprinkling here if FL |
| 14:16.02 | UnixDog | then again its summer in FL it rains every day |
| 14:16.16 | Mike8861 | i got a question, is there any mini-asterisk suitable to install on Nds ? |
| 14:16.17 | lmadsen | hugs Toronto's sunny, warm weather |
| 14:16.23 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:26.35 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:26.57 | Yourname`` | [TK]D-Fender: Still there? |
| 14:26.58 | keith4 | "Nds"? |
| 14:27.09 | [TK]D-Fender | your yes |
| 14:27.12 | Yourname`` | lmadsen: Short lived. Thunder tonight.. |
| 14:27.20 | Yourname`` | [TK]D-Fender: Great, starting the capture |
| 14:27.51 | [TK]D-Fender | Yourname``: Through tomorrow it seems |
| 14:28.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:28.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.38 | *** join/#asterisk emiller (n=ed@UNUSED-216-37-164-100.UNUSED.epix.net) |
| 14:32.37 | neurosys | Would 256 ram be enough just to play with asterisk? |
| 14:33.13 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:33.46 | tzafrir_laptop | neurosys, to play with it? yes |
| 14:34.08 | Yourname`` | I know eh.. :( |
| 14:34.15 | Yourname`` | It's the worst summer ever |
| 14:34.17 | Yourname`` | I'm sick cuz of it. |
| 14:34.19 | dominic1 | do you know any doc to ps converter? |
| 14:34.45 | tzafrir_laptop | openoffice: doc->pdf |
| 14:34.47 | neurosys | tzafrir_laptop: yeah.. just to play with a line or 2.. just to learn it. Im sure production is out of the question :) |
| 14:34.54 | Yourname`` | [TK]D-Fender: Got taken by others.. but now focusing on getting the debug in 2 mins |
| 14:35.27 | tzafrir_laptop | pdf->ps is trivial (pdftops or pdf2ps, one is from xpdf-utils, the other is from ghostscript) |
| 14:35.28 | dominic1 | I think antiword could help me |
| 14:35.36 | UnixDog | ok bbiab pain management time.. tap the knee drain it and then elctro threapy |
| 14:35.46 | [TK]D-Fender | dominic1: Any word processor with a PS driver :) |
| 14:36.19 | [TK]D-Fender | tasers UnixDog |
| 14:36.22 | *** join/#asterisk ManxPower (n=manxpowe@73.sub-70-222-63.myvzw.com) |
| 14:39.02 | tzafrir_laptop | abiword is nicer than openoffice for batch conversions. But less reliable. If antiword is good enough - all the better |
| 14:41.31 | dominic1 | I want to implement antiword with hylafax |
| 14:43.32 | [TK]D-Fender | dominic1: (.doc) -> E-mail -> PS -> Hylafax? |
| 14:44.09 | dominic1 | yes that's what I want |
| 14:44.42 | [TK]D-Fender | dominic1: more of a PITA for typicaly office use. Whats the selling point of this approach? |
| 14:45.04 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:45.57 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 14:46.28 | freezey | how would i set the phone to be able to login to the phone rather than setting the extensions statically? |
| 14:46.45 | dominic1 | users want to send attached word documents from their blackberry devices to customers fax |
| 14:47.00 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 14:47.54 | ManxPower | dominic1: And I would like to be able to do karate. But, much like an MS-Word/FAX gateway, it's more trouble than it's worth. |
| 14:48.43 | dominic1 | I would like to kick kungfupandas ass. I know what you mean... manx.... |
| 14:48.57 | [TK]D-Fender | ManxPower: Don't denigrate yourself with such a plebeian art ;) |
| 14:49.22 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:49.45 | ManxPower | dominic1: Are your users more technical than a turnip? Most word processors allow you to export to PDF, and PDF -> FAX is quite a bit easier. |
| 14:50.01 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:50.04 | *** join/#asterisk bkw_ (n=brian@70.234.164.166) |
| 14:50.05 | [TK]D-Fender | freezey: Consider your devices as "fixed", and all other processing is based on your dialplan. When somes dials "10", you go lookup which device it should send it to based on a "login". |
| 14:50.32 | [TK]D-Fender | ManxPower: He's talking FROM a CrackBerry |
| 14:50.36 | dominic1 | yes I know, but users are idiots and don't understand what I want to tell them. And the blackberry is not able to convet to pdf. :-( |
| 14:50.43 | [TK]D-Fender | ManxPower: Which frankly isn't the most flexible device ever. |
| 14:51.00 | ManxPower | dominic1: Best of luck. |
| 14:51.02 | [TK]D-Fender | dominic1: Your idea is fine... SAD, but fine. |
| 14:51.24 | [TK]D-Fender | dominic1: Makes me wonder what they'll want their little PDA to do next. |
| 14:51.24 | freezey | [TK]D-Fender: so in the beginning every phone must have an extension? i am not following... i want all phones to be able to be used on the fly nobody has a set phone so if they so happen to travel to another desk they can just login to that phone with their extension |
| 14:51.44 | *** join/#asterisk phpboy (n=shane@196.36.108.18) |
| 14:51.47 | phpboy | Jul 16 16:50:34 WARNING[4719]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'extensions-zero-prefix' |
| 14:51.54 | phpboy | what would that error generally mean? |
| 14:52.08 | [TK]D-Fender | freezey: An "extension" is a number in your dialplan. "phones" are "devices". THOSE are fixed. What happens when you dial a # is the job of your DIALPLAN. |
| 14:52.13 | ManxPower | phpboy: nobody entered anything in your IVR |
| 14:52.41 | ManxPower | freezey: What you want to do is VERY complex and requires significant understanding of Asterisk. |
| 14:52.43 | [TK]D-Fender | phpboy: And You didn't tell * what to do if noone did anything so it dropped them like a rock. |
| 14:52.44 | *** join/#asterisk javb (n=javb@96santrd02.codetel.net.do) |
| 14:52.57 | phpboy | I see |
| 14:53.09 | dominic1 | I have first the problem that faxmail is not converting correctly. My UTF-8 formatted outlook mails are not converted right. Do you know where I can adjust that? |
| 14:53.12 | [TK]D-Fender | freezey: Addendum : not that complicated, but if you can't envision it you're going to suffer. Hire it out. |
| 14:53.34 | ManxPower | dominic1: one of the problems is that I doubt many people can help you with the non-asterisk pieces |
| 14:53.36 | javb | hello, is there a way that i can play different files in a queue while waiting ? i mean, musiconhold > periodic announce 1 > musiconhold > periodic announce 2> ... and so on till answered ... |
| 14:53.53 | javb | So far, i just know a unique periodic anounce. |
| 14:53.56 | *** join/#asterisk jake[work] (n=jake[wor@99-205-43-237.area3.spcsdns.net) |
| 14:54.42 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 14:54.49 | putnopvut | javb: in Asterisk trunk and 1.6.0, you can specify a comma-separated list of files for periodic announcements. That's not possible in prior versions though. |
| 14:54.53 | Yourname`` | [TK]D-Fender: Got the debug, now formatting |
| 14:55.37 | ManxPower | Yourname``: why are you destroying information by reformatting? |
| 14:55.49 | javb | I see, what is Asterisk trunk ? |
| 14:56.10 | ManxPower | javb: The development version of Asterisk. |
| 14:56.27 | neurosys | Would anyone recommend a good stabe voip phone to learn aterisk with? |
| 14:56.29 | freezey | [TK]D-Fender: i would rather not hire it out i want to do it myself... what would i have to read in order to do this? |
| 14:56.39 | ManxPower | neurosys: Linksys |
| 14:56.49 | neurosys | Any linksys phone will do? |
| 14:56.55 | javb | putnopvut, are you totally sure there is no special way to do this ? ... i really need it, and i dont think it is viable to use Asterisk 1.6 ... |
| 14:57.01 | [TK]D-Fender | freezey: "core show applications" , "core show functions". |
| 14:57.01 | ManxPower | freezey: expect it to take a month or two. |
| 14:57.04 | freezey | [TK]D-Fender: i understand the dialplan thing you talked about earlier.. i just want people to be able to login to whatever pehone whenever they want.. i saw nortel had soething like this |
| 14:57.07 | javb | hence it is not stable. |
| 14:57.27 | [TK]D-Fender | freezey: And what you want has to be done at the dialplan level. |
| 14:57.31 | ManxPower | neurosys: most any linksys will work OK for you. I don't think I'd recommend it for busienss (I like Polycom for that) but the Linksys are inexpensive and work pretty well |
| 14:57.51 | putnopvut | javb: the only way I know to do that would be to either 1) backport the code in question from 1.6 to 1.4 or 2) make the Music on hold itself play various files |
| 14:58.10 | neurosys | ManxPower: thank you |
| 14:58.10 | freezey | [TK]D-Fender: ok and those core show applications and functions should point me in that direction? |
| 14:58.23 | [TK]D-Fender | neurosys: Polycom is the best quality phone out there, but linksys is decent. |
| 14:58.41 | neurosys | is voipsuppy a decent company? |
| 14:58.56 | [TK]D-Fender | freezey: as I said, what you need to do is entirely in the dialplan. Go learn what applications & functions are out there. |
| 14:58.59 | Yourname`` | [TK]D-Fender: I don't want to accidentally remove any important info you might need.. can I send it to you, and I can tell you what phone numbers are what please? |
| 14:59.09 | ManxPower | freezey: It's like you are asking how to write a CGI script for real time querying of airline flight info. What tools would you use for that? The answer is the list is so long I don't even want to think about it. |
| 14:59.10 | javb | putnopvut, ill stick with number 2 ;) |
| 14:59.11 | [TK]D-Fender | neurosys: overpriced. www.telephonydepot.com |
| 14:59.19 | neurosys | [TK]D-Fender: Thanks :) |
| 14:59.35 | freezey | ManxPower: well have you ever done something like what i am talking about? |
| 14:59.46 | [TK]D-Fender | Yourname``: Don't filter crap. I don't care who's doing what so long as I can trust what I see. |
| 14:59.59 | [TK]D-Fender | freezey: I have. |
| 15:00.01 | Yourname`` | [TK]D-Fender: Done good sir, I'm pbing it for you.. :) |
| 15:00.26 | ManxPower | freezey: you would use some sort of database (AsteriskDB, text file, other database) to keep track of what extension is associated with which device, then you need to write something to handle the login/logout. You should also somehow handle the situation when someone forgets to log out of their phone (which they will do all the time) |
| 15:00.30 | [TK]D-Fender | Yourname``: Do pull out completely irrelevent bits like calls I'm not caring about. |
| 15:01.10 | ManxPower | freezey: I spent about 10 days researching it. I determined that it was not worth the trouble. |
| 15:01.12 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:01.21 | freezey | ManxPower: yeah i figured out what to do about that whole login/logout if they forget... i was playing with the AutoCallbackLogin function and with my understanding that only logs them into a queue |
| 15:01.46 | [TK]D-Fender | freezey: Indeed. So forget those apps, they clearly aren't what you need. |
| 15:02.00 | freezey | ManxPower: the thing is is that these people are CSR reps and being able to bounce from station to station and having their extension follow them is what they truly need |
| 15:02.27 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 15:02.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:03.16 | ManxPower | freezey: WHY you want to do something so complex is not something I care about. |
| 15:03.44 | neurosys | wow. i can pickup a 2 line entry lvl polycom for 90 bucks. not bad |
| 15:04.20 | [TK]D-Fender | neurosys: + Power supply @ 15$ |
| 15:04.21 | freezey | ManxPower: i kind of felt like the way i originally explained it i might not have done it clear enough |
| 15:04.23 | [TK]D-Fender | +/- |
| 15:05.00 | neurosys | [TK]D-Fender: powersupply comes seperate? Oh .. i guess for people with PoE? |
| 15:05.20 | [TK]D-Fender | neurosys: Yup, they sell in a PoE expected environment. |
| 15:05.33 | [TK]D-Fender | neurosys: So either get a PoE injector, or a brick for it. |
| 15:05.34 | freezey | [TK]D-Fender: so which direction did you go to accomplish what i am looking to do? |
| 15:05.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:05.43 | neurosys | [TK]D-Fender: :) thx |
| 15:06.06 | [TK]D-Fender | neurosys: If you can get an injector close to the price of a brick, even if it is a little more, go for the injector. It'll be useful afterwards |
| 15:06.39 | [TK]D-Fender | freezey: All just AstDB in the dialplan. |
| 15:07.15 | freezey | [TK]D-Fender: kind of like what Manx refered to before? |
| 15:07.15 | neurosys | [TK]D-Fender: well.. im actually thinking.. because i plan to upgrade my switch. maybe i should just plan ahead and go with some PoE ports. |
| 15:07.16 | [TK]D-Fender | freezey: If you read all of *'s functions & apps and can't come up with this, then you don't seem to have the programmer mentality |
| 15:07.42 | [TK]D-Fender | neurosys: Describe the early phase and ending state of your purchase planning. |
| 15:07.45 | freezey | ok i will search through all of this atleast i know the base of what i am looking for |
| 15:07.51 | [TK]D-Fender | neurosys: Where are you going with *? |
| 15:08.09 | freezey | attending a conference this weekend that has a few blocks dedicated to asterisk should be pretty interesting |
| 15:08.24 | freezey | just thought to add that |
| 15:08.34 | neurosys | [TK]D-Fender: Well.. this is simply to learn about voip and asterisk. I have no official plans yet. I have several clients interested in voip setups so its skool time :) |
| 15:10.22 | [TK]D-Fender | neurosys: Honestly the Linksys is a LOT easier to configure, but less flexible because of it. A bit lower quality (build, audio), but not huge. I wouldn't suggest them for business in NA over Polycom but the learning experience is nice. Polycom is a lot more complex to configure, but the experience is very rewarding. |
| 15:13.05 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 15:13.23 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 15:14.24 | lmadsen | Yourname``: you're in downtown T.O. too right? |
| 15:14.34 | *** join/#asterisk jake[work] (n=jakepers@99-205-43-237.area3.spcsdns.net) |
| 15:17.19 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:18.04 | neurosys | [TK]D-Fender: well... im not known for taking the easy road ;) since my experience will be for businesses.. i might as well grind my teeth in. If theyre harder to configure, then learning that will make the linksys no trouble is i come across it. |
| 15:18.52 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 15:20.19 | [TK]D-Fender | neurosys: Good approach |
| 15:26.22 | *** join/#asterisk angryuser (n=dsfsdf@88.140.144.209) |
| 15:27.18 | angryuser | hello , is it possible to reload 'Dialplan' only? without touching and cuting all sip conversations |
| 15:27.52 | SuPrSluG | i'm trying to auto provision a polycom phone. is it possible without ever touching the polycom? |
| 15:28.38 | UnixDog | yes and its not easy |
| 15:28.44 | SuPrSluG | can dhcp tell the polycom to use ftp blah.blah files |
| 15:29.17 | [TK]D-Fender | angryuser: "reload" will not kill calls |
| 15:29.21 | UnixDog | there is a auto provisiong tool out there |
| 15:29.23 | SuPrSluG | UnixDog:can you point me int the right direction |
| 15:29.36 | UnixDog | that will setup the poly but I dont remember the name |
| 15:29.40 | UnixDog | let me look |
| 15:29.50 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:29.59 | [TK]D-Fender | SuPrSluG: Polycom uses FTP by default, and the user & pass are well documented as is the fact it pulls the server off option-66 |
| 15:30.27 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples/ |
| 15:31.04 | UnixDog | http://www.wintrisk.com/ppt.html |
| 15:31.43 | UnixDog | SuPrSluG: what firmware are you using |
| 15:32.47 | SuPrSluG | differ depending on the phone. 500,501,600.601 |
| 15:32.48 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 15:33.42 | *** join/#asterisk TraydenT (n=traydent@212.73.216.2) |
| 15:33.54 | UnixDog | well the current is 3.0.3 |
| 15:34.05 | SuPrSluG | [TK]D-Fender:that's what i'm trying to accomplish. doess option 66 allow for ftp? or just tftp? |
| 15:34.10 | UnixDog | `for all currently supported phones |
| 15:34.37 | UnixDog | there is a script that uploads a file to the phone |
| 15:34.41 | UnixDog | I am hunting for it |
| 15:34.45 | UnixDog | just chill |
| 15:34.47 | SuPrSluG | 3.0.3 only add fixes for the conference phones. otherwise it's the same as 3.0.2 |
| 15:35.08 | SuPrSluG | k |
| 15:35.32 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 15:35.39 | [TK]D-Fender | SuPrSluG: The Polycom will use that IP for FTP as thats its default protocol |
| 15:36.20 | Yourname`` | lmadsen: Yeah, smackdab |
| 15:36.41 | lmadsen | Yourname``: I just bought a place at Yonge and Dundas... you near there? |
| 15:36.45 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:36.53 | Yourname`` | lmadsen: Right there.. where you at? |
| 15:37.04 | lmadsen | Pantages Towers |
| 15:37.07 | lmadsen | Tower* |
| 15:37.44 | lmadsen | Yourname``: phone! brb |
| 15:37.52 | Qwell | lmadsen: it's him - stalker |
| 15:38.37 | Yourname`` | haha |
| 15:38.43 | jbeez | I was at Dundas square like last month |
| 15:38.59 | Yourname`` | In the elevator "So, leif.. whats with the 1.6 commit from last night?" |
| 15:38.59 | Qwell | party at lmadsen |
| 15:39.00 | Yourname`` | lol |
| 15:39.01 | Qwell | 's |
| 15:39.36 | jbeez | I was visiting toronto for business and went down there to grab some food, ended up going into that place with the movie theater and they had a little food court type thing |
| 15:39.46 | UnixDog | its a endpoint manager app + scripts |
| 15:40.04 | ManxPower | I love Toronto |
| 15:40.41 | Yourname`` | Yeah, that's the new Toronto life mall |
| 15:40.47 | dominic1 | I installed hylafax from source, now nothing is in /etc/hylafax |
| 15:40.50 | [TK]D-Fender | ManxPower: its a nice place to come from ;) |
| 15:40.51 | dominic1 | is that correct? |
| 15:41.00 | jbeez | I'm from philly, and I thought toronto was awesome, I'd consider moving there |
| 15:41.07 | Yourname`` | It's stupid in a way cuz it's filled with students from the university that uses the movie theatres as lecture halls sometimes |
| 15:41.19 | Yourname`` | jbeez: Don't move here. Summer is the only fun time :( |
| 15:41.20 | [TK]D-Fender | dominic1: configs don't go there... hylafax is wierd that way |
| 15:41.35 | jbeez | Yourname``: I wouldn't be able to import my car there, so I probably wouldn't anyway |
| 15:42.25 | dominic1 | But I got this error: /etc/hylafax/typerules: Can not open type rules file. |
| 15:43.47 | test333 | has anybody got an idea on how to transfer a called channel to a different context? |
| 15:44.13 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 15:44.22 | Yourname`` | jbeez: Yeah, and to be able to use it is another thing ;) |
| 15:44.35 | test333 | sorry if my asterisk vocabulary isn't very precise |
| 15:44.39 | test333 | what do you mean? |
| 15:47.32 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:47.58 | [TK]D-Fender | test333: What is on each end? |
| 15:48.50 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279395306.dsl.bell.ca) |
| 15:48.51 | UnixDog | ok no luck so far on the polycom endpoint manager |
| 15:48.55 | mandd | Heya! |
| 15:48.59 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 15:49.13 | mandd | I got voip > asterisk > lynksys pap2 > FAX |
| 15:49.24 | mandd | and its now working |
| 15:49.41 | mandd | I mean fax calls, voice etc works, but as soon as it get to actual faxing, it hangs up/fails |
| 15:49.48 | mandd | on both, sending and receiveing |
| 15:50.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:50.46 | *** join/#asterisk tomo_connor (n=toconnor@rabbit.dbplc.com) |
| 15:51.54 | UnixDog | the only endpoint manager I find is for trashbox |
| 15:52.33 | [TK]D-Fender | UnixDog: Easy to make your own. |
| 15:52.34 | Hertzy3 | I am having problems installing asterisk on a CentOS machine. Can someone help me out? |
| 15:52.45 | [TK]D-Fender | Hertzy3: What knd of problems? |
| 15:53.02 | Hertzy3 | Ive downloaded asterisk, zaptel, libpri, and addons |
| 15:53.21 | UnixDog | br |
| 15:53.24 | UnixDog | b |
| 15:53.27 | Hertzy3 | I first tried to install zaptel. I did ./configure which works. but I get errors when I do make install |
| 15:53.48 | Hertzy3 | I checked the readme, I have all of the necessary components, and all up to date kernels |
| 15:54.02 | outtolunc | did you check the topic? |
| 15:54.09 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 15:54.38 | Hertzy3 | oh wow |
| 15:54.45 | Hertzy3 | no i didnt, but I have now |
| 15:54.58 | outtolunc | note: that applies to anyone doing 5.x yum updates also |
| 15:55.13 | [TK]D-Fender | ~centos52bug |
| 15:55.14 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 15:55.30 | Hertzy3 | ~centos52bug |
| 15:55.30 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 15:55.35 | [TK]D-Fender | Hertzy3: Just manually desect XPP in zaptel's make menuconfig and you should be fine |
| 15:55.57 | jaytee | Hertzy, you just need to rerun make clean, ./configure and then make menuselect and deselect the Xorcom Astribank driver and then you can compile and install. |
| 15:56.12 | [TK]D-Fender | jaytee: No prize for 3rd place :) |
| 15:56.13 | jaytee | damn, [TK]D-Fender beat me to it. |
| 15:56.15 | Hertzy3 | thank you guys very much |
| 15:56.17 | jaytee | hahaha |
| 15:56.28 | jaytee | [TK]D-Fender, hit that bug yesterday or the day before |
| 15:56.46 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:56.51 | [TK]D-Fender | jaytee: and I put it in the topic almost 3 weeks ago :) |
| 15:59.37 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279395306.dsl.bell.ca) |
| 16:01.26 | *** join/#asterisk sack (n=sack@70.Red-88-24-156.staticIP.rima-tde.net) |
| 16:03.50 | dominic1 | one more question: when using fax2mail with utf-8 formated mail german umlauts are not working. Any ideas? |
| 16:04.46 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 16:06.15 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 16:06.16 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 16:10.37 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 16:11.49 | tzafrir_laptop | dominic1, incorrectly-formatted mail? |
| 16:12.23 | dominic1 | ähm sorry I was wrong I meant mail2fax |
| 16:12.34 | dominic1 | if I send UTF-8 mails to a fax |
| 16:12.55 | dominic1 | the characters in the fax aren't correct |
| 16:13.40 | dominic1 | I searched with google and the mailinglists but I think faxmail isn't supporting utf8 |
| 16:14.28 | coppice | how are you converting the mail to a FAX? |
| 16:15.19 | mandd | what I want the actual fax machine to work, no emails, just good old FAx, as in VOIP > asterisk > pap2 lynksys > FAX machine |
| 16:15.23 | mandd | can that be done? |
| 16:15.44 | dominic1 | with a executable called faxmail |
| 16:16.57 | coppice | where did the executable come from? there appear to be various things called faxmail |
| 16:17.30 | dominic1 | /usr/local/bin/faxmail |
| 16:17.39 | *** join/#asterisk ToTo (n=ToTo@209.8.41.200) |
| 16:17.46 | coppice | duh! |
| 16:18.03 | dominic1 | the executable is from hylafax |
| 16:19.48 | *** join/#asterisk chinku (n=chinku@cpe-72-225-237-124.nyc.res.rr.com) |
| 16:20.14 | coppice | I don't think that one currently supports UTF-8. It treats everything as ISO-8859-1 |
| 16:21.02 | dominic1 | okay thank you |
| 16:21.58 | [TK]D-Fender | mandd: Fax over VoIP = real trouble |
| 16:23.01 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:23.22 | dominic1 | thank you guys, see you tomorrow |
| 16:25.05 | caio1982 | are astricon sessions recorded in video, streamed or something? |
| 16:27.57 | javb | when i press i key, during play of my file using background, example "press 1 fro ..." when 1 is pressed it take like 2-3 seconds to execute the exten, any idea? |
| 16:29.54 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 16:30.08 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:31.19 | *** join/#asterisk MrNaz (n=naz@ppp59-167-174-142.lns1.mel4.internode.on.net) |
| 16:31.59 | ManxPower | javb: what other stuff do you have that begins with 1 |
| 16:32.13 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 16:33.34 | rickross | anyone know how to obtain and build the new DAHDI piece required to make MeetMe work under the current 1.6 svn trunk? |
| 16:34.05 | ManxPower | rickross: they might know on #asterisk-dev |
| 16:35.01 | Corydon76-dig | rickross: svn co http://svn.digium.com/svn/dahdi/complete |
| 16:35.40 | Corydon76-dig | err, svn co http://svn.digium.com/svn/dahdi/complete/trunk dahdi-trunk |
| 16:36.22 | mandd | Unsupported SDP media type |
| 16:36.22 | rickross | ahh, thx! |
| 16:36.25 | mandd | when I try to fax |
| 16:36.26 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 16:36.35 | rickross | I was confused because there was no trunk under dahdi |
| 16:36.53 | tzafrir_laptop | Corydon76-dig, http://svn.digium.com/svn/dahdi/linux-complete/ ? |
| 16:37.11 | Corydon76-dig | tzafrir_laptop: no |
| 16:37.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.92.35) |
| 16:37.57 | tzafrir_laptop | svn: URL 'http://svn.digium.com/svn/dahdi/complete/trunk' non-existent in that revision |
| 16:38.37 | Corydon76-dig | Oh, has it moved? |
| 16:38.59 | *** join/#asterisk XnOSX (n=XnOSX@212.145.170.219) |
| 16:39.00 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 16:39.10 | rickross | well, most of the major items in the svn repo have a trunk directly under them |
| 16:39.23 | *** part/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 16:41.22 | ManxPower | mandd: that has got to be one of the worst pleas for help I've seen here in a long time. PASTEBIN the failed call. |
| 16:41.58 | *** join/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 16:42.51 | ManxPower | mandd: Of course, there is an EASY to make sure your faxes work. Don't run them thru asterisk. |
| 16:43.22 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 16:43.36 | rickross | excellent, now dahdi built and asterisk allows me to build MeetMe |
| 16:43.39 | rickross | thank you |
| 16:43.45 | silvertip257 | I'm trying to connect an Ekiga softphone to an Asterisk PBX I have set up - Ekiga says it fails registration and nothing shows up in my debug session in the Asterisk CLI. Please give me some tips. Thanks. |
| 16:45.42 | rickross | if we enable chan_ooh323, would that mean we could connect with Skype users? |
| 16:46.07 | ManxPower | rickross: why in the WORLD would you think that. |
| 16:47.44 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 16:47.48 | rickross | maybe I am a dimwit, ManxPower |
| 16:48.03 | rickross | but, in fact, I did think that :) |
| 16:48.09 | ManxPower | rickross: I'm sure you didn't pull the idea out of thin air? |
| 16:48.24 | rickross | I guess someone had suggested to me that Skype is using H.323 |
| 16:48.42 | ManxPower | But the question demonstrates a scary lack of understanding of H323 or Skype. |
| 16:48.50 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:48.59 | rickross | ManxPower - that may be true |
| 16:49.10 | ManxPower | ~skype |
| 16:49.10 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 16:49.29 | shido6 | in general..... heh |
| 16:49.34 | rickross | another possible explanation is that it is tangential to what I really do, and I am lazy and optimistically hoping to learn "just enough" to get itoperational |
| 16:49.34 | shido6 | it can be done |
| 16:50.00 | shido6 | you'll just learn you need to do more reading |
| 16:50.04 | rickross | shido6, so it isn't worth the hassle? |
| 16:50.10 | ManxPower | rickross: and that is why you'll fail. |
| 16:50.16 | shido6 | sure it is, if you're getting something out of it worth YOUR time |
| 16:50.26 | rickross | Manx, thanks for the guidance :) |
| 16:50.35 | CVirus | Zap/2-1 .. what does -1 stand for ? |
| 16:50.56 | shido6 | I want a pb&j sandwich right now, do I want to put in the work to build one, or the work to motivate someone else to do it for me. |
| 16:50.59 | ManxPower | CVirus: it stands for first call -2 would stand for 2nd call (like in a call waiting thing) |
| 16:51.20 | CVirus | thanks |
| 16:51.29 | CVirus | what is the variable for the channel name only .. without the call |
| 16:51.38 | CVirus | like Zap/1 or Zap/2 |
| 16:51.53 | ManxPower | there is none, but the info is contained in CHANNEL |
| 16:52.02 | CVirus | okay thanks |
| 16:52.53 | *** join/#asterisk angryuser (n=dsfsdf@88.140.144.209) |
| 16:54.50 | mandd | I can send fax. but cant receive |
| 16:54.58 | mandd | it connects and after claims that extension is busy |
| 16:55.01 | *** join/#asterisk nipsy (n=nipsy@207.200.46.196) |
| 16:55.02 | mandd | but its not |
| 16:55.05 | mandd | Unsupported SDP media type |
| 16:55.11 | mandd | error on send adn receive |
| 16:55.16 | mandd | but it still sends |
| 16:55.35 | silvertip257 | ManxPower: I have asterisk on a WRT54 router - the port 5060 for sip doesn't appear to be open, so my softphones aren't able to register. Any ideas? Thanks. |
| 16:55.48 | ManxPower | mandd: I can't help yo because you did not provide the information I requested 12 mins ago |
| 16:56.08 | mandd | sorry ManxPower, I missed that, will go read. |
| 16:56.49 | *** join/#asterisk emiller (n=ed@UNUSED-216-37-164-100.UNUSED.epix.net) |
| 16:57.21 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 16:59.13 | *** part/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:59.51 | silvertip257 | ManxPower: Wireshark shows my softphone sending SIP registration requests, but the PBX sending ICMP pkts back |
| 17:00.14 | mandd | ManxPower http://pastebin.com/m47ae1646 |
| 17:00.14 | ManxPower | silvertip257: Did I volunteer to help you? |
| 17:00.20 | silvertip257 | fine |
| 17:00.22 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 17:00.41 | mandd | and I configured PAP2 follwing http://www.fixya.com/support/t277432-fax_thru_pap2 |
| 17:00.46 | mags2 | sudo make me a sammich |
| 17:01.00 | mandd | so its voip -> asterisk -> pap2 -> FAX |
| 17:01.05 | ManxPower | mandd: what version of Asterisk are you using? |
| 17:01.28 | mandd | fax sends just fine, but wont receive, Asterisk 1.4.20.1 |
| 17:01.54 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 17:01.56 | ManxPower | mandd: well you are trying to use T.38. I can't help with T.38 problems. |
| 17:02.11 | [TK]D-Fender | ManxPower: Down boy! |
| 17:02.28 | mandd | oh |
| 17:02.32 | ManxPower | [TK]D-Fender: Well I can't help with T.38 problems. |
| 17:02.47 | mandd | I can just get asterisk to forward whatever comes it's way on that number |
| 17:02.52 | mandd | without asking? |
| 17:03.28 | [TK]D-Fender | ManxPower: I'm talking about silvertip257 whose head you bit nearly clean off... |
| 17:03.48 | mandd | essentially all it was to do is just stream that noise |
| 17:03.54 | mandd | and let the fax figure it out |
| 17:04.35 | coppice | ManxPower: he's not using T.38 if he's using a PAP2 |
| 17:06.26 | *** join/#asterisk salmander (n=omi@CPE0002b323839e-CM001225024984.cpe.net.cable.rogers.com) |
| 17:06.28 | freezey | [TK]D-Fender: ok so let me ask you this... is it possible to rebuild a sip.cfg and let the phone pull it down and setup a login function like that and ahve it pull from a database? |
| 17:06.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:06.56 | salmander | Hello. Can someone suggest a good voip service in canada? I would like to buy a pool of minutes and buy DIDs that will share those minutes. |
| 17:07.11 | salmander | Looking for 416/647 and 905 DIDs |
| 17:07.34 | mandd | and my * keeps doing this: http://pastebin.com/m7af3f5b1 |
| 17:07.44 | mandd | while both registers are registered fine |
| 17:08.06 | ManxPower | coppice: http://www.mail-archive.com/astlinux-users@lists.kriscompanies.com/msg00244.html |
| 17:08.15 | ManxPower | SOMEONE is trying to use T.38 |
| 17:08.21 | [TK]D-Fender | freezey: Not viable. yes you can perform the SIP.CONF updates, but you won't force the phone to reregister and update the point to call at. Its also a boatload of scripting. |
| 17:08.41 | ManxPower | freezey: phones do not pull their config from Asterisk!!! |
| 17:08.44 | [TK]D-Fender | salmander: lest.net |
| 17:09.18 | mandd | but * doesnt have to understand t.38 I just want it to pass the signal on to fax |
| 17:09.23 | coppice | ManxPower: what does that old mail have to do with someone trying to use a PAP2? |
| 17:09.32 | salmander | lest.net is some restaurant |
| 17:09.35 | mandd | can I somehow convince it to ignore what it is, and justdo it? |
| 17:09.37 | salmander | I'm not hungry for food |
| 17:09.38 | ManxPower | coppice: this PAP2 user is getting that message. |
| 17:09.40 | [TK]D-Fender | freezey: please understand that this is largely a wrong approach. ManxPower and I have both told you which way you should be starting on this. |
| 17:09.41 | freezey | ManxPower: what i am saying is setup a tftp server and have custom configs for the phone to pull that has it look at the asteriskbox db and check for the AgentID and PW you enter at login |
| 17:09.51 | [TK]D-Fender | salmander: les.net * |
| 17:09.55 | freezey | i know i ma just thinking of other wyas |
| 17:09.57 | freezey | ways** |
| 17:10.02 | salmander | thanks |
| 17:10.56 | coppice | ManxPower: but the message has nothing to do with T.38 |
| 17:11.08 | ManxPower | coppice: what does it have to do with? |
| 17:11.11 | *** join/#asterisk freddyk (n=freddy@host104-23-dynamic.42-79-r.retail.telecomitalia.it) |
| 17:11.22 | ManxPower | coppice: this is what mandd posted http://pastebin.com/m7af3f5b1 |
| 17:11.27 | coppice | Its a telephony signaling event, I expect, |
| 17:11.35 | ManxPower | argh! |
| 17:11.40 | mandd | thats a wrong one :) |
| 17:11.41 | ManxPower | (12:00:14 PM) mandd: ManxPower http://pastebin.com/m47ae1646 |
| 17:11.42 | ManxPower | there! |
| 17:11.50 | mandd | speaking of which |
| 17:11.53 | freddyk | Anyone knows if asterisk 1.6 supports Comfort Noise Generation ? |
| 17:11.54 | mandd | is it suppoused to do that? |
| 17:11.57 | mandd | over and over again? |
| 17:12.04 | mandd | like every second? |
| 17:12.08 | ManxPower | do what? |
| 17:12.14 | ManxPower | either start being specific or give up |
| 17:12.18 | freddyk | ~centos52bug |
| 17:12.19 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 17:12.19 | mandd | # |
| 17:12.19 | mandd | Really destroying SIP dialog '77dd91712f84c3153143c9b45420e696@192.168.2.30' Method: REGISTER |
| 17:12.19 | mandd | # |
| 17:12.19 | mandd | REGISTER attempt 1 to 4168491130@tor3.voipportal.ca |
| 17:12.23 | mandd | that |
| 17:12.50 | mandd | i am not even sure what it means |
| 17:13.01 | mandd | almost as if it re-registers every second |
| 17:13.36 | ManxPower | mandd: I cannot help your further. |
| 17:13.45 | CVirus | how can I force asterisk to end a call on a Zap channel after the other party have ended it ? |
| 17:13.55 | shido6 | Hangup ? |
| 17:13.56 | ManxPower | freddyk: It it does I'll bet it will be listed in the changelog |
| 17:14.15 | coppice | mandd: if you add T.38 passthrough support to your asterisk it won't help. The PAP2 doesn't support T.38 |
| 17:14.33 | mandd | coppice but its send fax out |
| 17:14.36 | mandd | just fine |
| 17:14.59 | mandd | fax -> pap2 -> asterisk -> regular fax number WORKS |
| 17:15.14 | coppice | maybe, but not by T.38. If you are lucky FAX will often work over VoIP. don't bet on it, though |
| 17:15.46 | freddyk | ManxPower: where i can read it ? |
| 17:16.04 | [TK]D-Fender | freddyk: In your source tarbal. |
| 17:16.10 | [TK]D-Fender | freddyk: Read the docs it comes with |
| 17:16.26 | ManxPower | there's a whole DIRECTORY of docs in the asterisk source code |
| 17:16.38 | mandd | why does it keep on saying REGISTER attempt 1 on a on again |
| 17:16.43 | mandd | when it is already registered |
| 17:16.53 | mandd | if I do sip show registry |
| 17:17.00 | mandd | both numbers are registered |
| 17:17.18 | mandd | Really destroying SIP dialog , REGISTER attempt 1 to ..... |
| 17:17.21 | mandd | keeps on going |
| 17:17.22 | anonymouz666 | FAX sucks. How can you trust in something that sometimes....can... work... I wonder how I'd say that to my customers |
| 17:17.23 | [TK]D-Fender | mandd: pastebin your sip debug. That useless notice is BS |
| 17:17.30 | ManxPower | mandd: sip show registry shows servers you are registered TO. |
| 17:17.48 | ManxPower | NOT incoming registers. For that "sip show peers" |
| 17:18.02 | freddyk | [TK]D-Fender: in 1.6 beta 9 tarball, into the file named CHANGES there is no track of comfort noise generation support. Is it a trunk feature ? |
| 17:18.03 | mandd | oh, so I can just ignore that? |
| 17:18.21 | [TK]D-Fender | freddyk: If you don't see it, chances are it doesn't exist. |
| 17:18.30 | ManxPower | freddyk: 1.6 pretty much IS trunk. Don't expect CNG in Asterisk |
| 17:18.36 | mandd | [TK]D-Fender verbose 10? |
| 17:18.43 | ManxPower | If you need CNG, then Asterisk will not work for you. |
| 17:18.52 | [TK]D-Fender | mandd: Verbose is NOT SIP DEBUG |
| 17:19.02 | mandd | err, sorry, |
| 17:19.19 | Corydon76-dig | "sip set debug on" |
| 17:19.50 | Katty | wibbles. |
| 17:20.22 | [TK]D-Fender | Katty may wobble but she won't fall down! |
| 17:21.36 | freddyk | ManxPower: So there is no chance to ever get this feature in asterisk 1.6 ? |
| 17:22.09 | ManxPower | freddyk: Maybe if you want to write it, but it requires significant redesign of big parts of Asterisk (or at least the RTP stuff) |
| 17:22.14 | mandd | [TK]D-Fender so there is nothign I can do about incoming fax? |
| 17:22.16 | anonymouz666 | Corydon76-dig: how would be the correct way to get two values or more in the same select using func_odbc? |
| 17:22.52 | ManxPower | mandd: sure there is, but you are not even close to even understanding what the issue is. |
| 17:23.09 | ManxPower | your incoming faxes are using T.38. Make them stop. |
| 17:23.24 | mandd | by configuring * ? |
| 17:23.32 | ManxPower | no. |
| 17:23.42 | ManxPower | the call is T.38 already when it comes in. |
| 17:23.44 | mvanbaak | func_odbc...... I needed to test something with that |
| 17:23.47 | [TK]D-Fender | mandd: You keep throwing random unqualified comments around like a psycho girlfriend who keeps asking "Don't you love me any more?" out of the blue in the middle of a happy gathering. |
| 17:23.49 | mvanbaak | AH |
| 17:23.51 | ManxPower | so what would be the logical place to fix that? |
| 17:24.00 | mvanbaak | now I remember |
| 17:24.37 | mandd | on a VOIP level? |
| 17:24.37 | CVirus | can I compare strings in asterisk ? like exten => _X.,1,Verbose($[${CHANNEL:0:3}="SIP"]) |
| 17:25.28 | mandd | ManxPower if someone sends me fax, from their fax machine, it is going to be t.38 no? |
| 17:25.43 | mandd | i can't control that |
| 17:26.14 | mandd | [TK]D-Fender, don't you love me any more? |
| 17:26.43 | [TK]D-Fender | mandd: No, now get lost psycho! :p |
| 17:27.05 | mandd | damn, i saw that one coming |
| 17:27.26 | [TK]D-Fender | CVirus: Yes you can, and you are clsoe but there is a bug in there due to your lack of understanding on *'s use of variables |
| 17:28.16 | *** join/#asterisk MissNeBuN (i=hidden-u@gw.mypublisher.com) |
| 17:28.44 | Corydon76-dig | anonymouz666: two arguments or two values? |
| 17:28.58 | anonymouz666 | two values |
| 17:29.03 | Corydon76-dig | In 1.4 or 1.6? |
| 17:29.06 | anonymouz666 | 1.4 |
| 17:29.26 | Corydon76-dig | Set(ODBC_...(...)=123\,456) |
| 17:29.43 | CVirus | [TK]D-Fender: exten => _X.,1,Verbose($["${CHANNEL:0:3}"="SIP"]) |
| 17:29.43 | Corydon76-dig | In 1.6, omit the backslash |
| 17:29.45 | CVirus | [TK]D-Fender: :-) |
| 17:29.55 | [TK]D-Fender | CVirus: Much better :) |
| 17:30.18 | mvanbaak | omg, func_odbc rox |
| 17:30.31 | CVirus | [TK]D-Fender: I would really love to show you my dial plan after I'm done with it ... the book and the wiki helped me so much |
| 17:30.36 | [TK]D-Fender | CVirus: Feel free to engage in a 1-line rant about *'s lack of variable type treatment :) |
| 17:30.48 | CVirus | hehe |
| 17:31.17 | mandd | can I engage in a 1-line rant about *'s lack of t.38 ? |
| 17:31.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:31.25 | CVirus | asterisk is a dynamically typed language :-p |
| 17:32.20 | rwaite | you know what kinda cool about asterisk? i'll spend two days trying to fix a bug and then realize most of the problem was not rtfming, then rtfm, and its two lines in a conf file |
| 17:32.38 | [TK]D-Fender | mandd: 1.6 does. So wait around, get an endpoint that does, or stop whining. |
| 17:33.02 | rwaite | is there a roadmap for 1.6's release |
| 17:33.02 | mandd | <PROTECTED> |
| 17:33.11 | [TK]D-Fender | mandd: PAP2 |
| 17:33.28 | [TK]D-Fender | mandd: What the hell does an analog fax machine know about VoIP? |
| 17:33.44 | mandd | so when I send out it uses a compatible codec? |
| 17:33.47 | mandd | thats why it works? |
| 17:33.49 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 17:33.57 | mandd | but on incoming, its t.38 ? |
| 17:34.52 | rwaite | lol |
| 17:36.28 | [TK]D-Fender | mandd: on outgoing it doesn't whine because the PAP2 isn't trying to offer T.38 and your ITSP doesn't care. On incoming THEY know its a fax and try to force T.38 |
| 17:36.53 | mandd | [TK]D-Fender willl this work: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 |
| 17:37.16 | *** join/#asterisk Rico29 (n=Rico@nor75-29-88-188-75-101.fbx.proxad.net) |
| 17:37.22 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:38.10 | mandd | or i'll end up recompiling my * unbootable? |
| 17:38.44 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 17:41.25 | [TK]D-Fender | mandd: Let me make this very clear : the PAP2 does NOT SUPPORT T.38. Pass-though doesn't mean @&$! when your ENDPOINT doesn't support it. |
| 17:41.51 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 17:42.11 | mandd | oh |
| 17:42.12 | mandd | yeah |
| 17:42.22 | mandd | sorry [TK]D-Fender |
| 17:42.27 | mandd | i wasnt thinking straight |
| 17:42.31 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 17:42.39 | CVirus | if i understand correctly ... channel variables are used within the same context but global variables can be used within any context ? |
| 17:42.42 | CVirus | am I wrong ? |
| 17:42.55 | [TK]D-Fender | mandd: And please note the glorious absence of the PAP2 from the giant "compatibility list" on that page. |
| 17:42.57 | doolph | question: can I install Asterisk without zaptel? because I don't have any hardware, it will be used as sip registrar |
| 17:43.15 | mvanbaak | CVirus: correct |
| 17:43.16 | rwaite | yes, but no. you shouldnt |
| 17:43.25 | rwaite | it uses the zap driver as a timing source |
| 17:43.28 | [TK]D-Fender | CVirus: No. Global variables are shared between all CALLS. |
| 17:43.35 | doolph | umm |
| 17:43.46 | mandd | PAP2T |
| 17:43.48 | doolph | so I have to install zaptel then compile dummy driver? |
| 17:43.49 | mvanbaak | rwaite: that only matters when you want meetme or iax2 trunking |
| 17:43.50 | mandd | is my answer |
| 17:43.52 | *** join/#asterisk originaljimdandy (n=jim@s66-76-129-58.mthmcmta01.mthoar.lt.sta.suddenlink.net) |
| 17:43.53 | CVirus | [TK]D-Fender: channel variables can be shared in different contexts ? |
| 17:43.57 | [TK]D-Fender | CVirus: Channel variables are limited to the scop of the channel (call). |
| 17:44.12 | rwaite | mvanbaak: ahh. well i'm doing iax2 trunking so maybe thats why i had it set in my head |
| 17:44.19 | [TK]D-Fender | CVirus: in programming terms, all channel variables are "global variables" within the scop of the call. |
| 17:44.29 | doolph | hello |
| 17:44.32 | CVirus | [TK]D-Fender: got that .. thanks |
| 17:44.38 | outtolunc | don't forget about _ and __ |
| 17:44.51 | mandd | whoa! |
| 17:44.55 | mandd | i have pap2t |
| 17:44.55 | [TK]D-Fender | CVirus: But an Asterisk Global Variable is a variable that if you change in one call will exist and contain that "last changed" value from before. |
| 17:45.07 | mandd | [TK]D-Fender my endpoint does in fact support t.38 |
| 17:45.08 | CVirus | sure |
| 17:45.22 | [TK]D-Fender | outtolunc: Yes, standard channel variable inheritance stil reigns supreme :) |
| 17:45.41 | doolph | does asterisk only SIP registrar need ztdummy?? |
| 17:45.53 | [TK]D-Fender | mandd: that has been a hot topic and you can feel free to waste more time and try taking that up with coppice. |
| 17:45.58 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 17:45.59 | [TK]D-Fender | mandd: Good luck with all that. |
| 17:46.00 | hsv-al | . |
| 17:46.11 | [TK]D-Fender | doolph: No. |
| 17:46.11 | Katty | oh |
| 17:46.13 | Katty | matt's not here |
| 17:46.17 | Katty | anyone seen hmmhesays lately? |
| 17:46.21 | mandd | :), i'll just mess with it myself |
| 17:46.25 | doolph | ok |
| 17:46.28 | mandd | thanks for help [TK]D-Fender ! |
| 17:46.32 | coppice | mandd: you said you endpoint is a PAP2. they do not support T.38 |
| 17:46.38 | [TK]D-Fender | Katty: Been a week or two. |
| 17:46.41 | mandd | and ManxPower and coppice |
| 17:47.07 | mandd | I am googling PAP2T t.38 |
| 17:47.14 | anonymouz666 | coppice: what about the latest firmware with T.38 options? |
| 17:47.16 | [TK]D-Fender | coppice: And the last state we got in here concerning the PAP2T was not really T.38 either. |
| 17:47.17 | mandd | and forums claim that it does suport it |
| 17:47.34 | [TK]D-Fender | coppice: (I was just just lurking and reading while you were the active participant) |
| 17:48.11 | coppice | forums claim all sorts of things. Linksys claim is doesn't. the manual makes not mention of it. the configuration menus don't mention it. the PAPA2T is the same |
| 17:48.31 | mandd | damn |
| 17:48.43 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:49.27 | nny_1 | so by magically reading the docs I found that you can force new voicemail users to set a greeding and password (awesome). |
| 17:49.42 | nny_1 | is there a way for them to change the password once it |
| 17:49.43 | rwaite | yes that is awesome |
| 17:49.46 | nny_1 | is set? |
| 17:49.52 | nny_1 | i checked advanced options etc |
| 17:49.56 | rwaite | yes, its in the docs |
| 17:50.02 | rwaite | 0 for mailbox options |
| 17:50.03 | nny_1 | kk i'll look more |
| 17:50.06 | nny_1 | thanks |
| 17:50.07 | coppice | a number people say the big upgrade in the PAP2T is the addition of T.38 support. well, try to find it in the menus. The SPA 2102 and the SPA 3102 do support T.38 |
| 17:50.47 | nny_1 | rwaite: is there a better source for info than voip-info? Like an official man page or the like? |
| 17:51.02 | rwaite | nny: the doc subdir of the asterisk sources |
| 17:51.37 | rwaite | the wiki is 99% rehashes of that directory |
| 17:51.51 | rwaite | (in my experience, and the included docs are much clearer) |
| 17:52.11 | [TK]D-Fender | nny_1: Hidden in the big print again ;) |
| 17:52.45 | nny_1 | rwaite: thanks |
| 17:53.11 | nny_1 | [TK]D-Fender: heh |
| 17:57.30 | CVirus | will this conditional branching execute the two lines after it ? http://rafb.net/p/GBgN5d43.html |
| 17:57.33 | CVirus | or only the first one ? |
| 17:57.54 | CVirus | if it's true I mean |
| 18:01.30 | mags2 | CVirus normally in that case both lines will execute, but with both label as Zap1 I have no idea what will happen...! |
| 18:05.47 | [TK]D-Fender | CVirus: You have repeated a priority tag. That is a big no-no. |
| 18:05.59 | [TK]D-Fender | CVirus: And you don't seem to have a Zap2 tag anywhere |
| 18:06.11 | CVirus | there's a zap2 tag but i just didn't paste it |
| 18:06.28 | [TK]D-Fender | CVirus: point #1 still important |
| 18:06.53 | CVirus | ok .. thanks |
| 18:07.01 | [TK]D-Fender | CVirus: Dialplan still executes consecutively through "n". Don't repeat labels |
| 18:07.16 | CVirus | ok |
| 18:07.28 | [TK]D-Fender | CVirus: otherwise the 2nd one might override the first and that one won't get executed |
| 18:07.38 | CVirus | oops |
| 18:12.05 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 18:12.23 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 18:14.12 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-018-148.mycingular.net) |
| 18:14.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:15.23 | *** part/#asterisk Maxous (n=Maxous@mobile-166-214-018-148.mycingular.net) |
| 18:16.03 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 18:16.35 | moy | korihor: thanks for the patches :) ... I will review them tonight and commit if OK, I suppose you tested those in Venezuela? |
| 18:17.16 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:19.03 | tzanger | network connectivity dies too apparently |
| 18:19.23 | hsv-al | 4 days ive had my iphone |
| 18:19.25 | x86 | ManxPower: well, it wasn't a signalling issue... I changed all spans from kewlstart to loopstart, no callprogress, no busydetect, people are still getting random dropped calls and the "robotic voice" distortion |
| 18:19.27 | hsv-al | and still no sip client, bleh |
| 18:20.11 | coppice | moy: what's the strangest place you've had people ask about for R2? |
| 18:22.20 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 18:23.30 | x86 | anyone have any ideas on random dropped calls from an analog phone to the outside world over E&M Wink CAS T1? |
| 18:23.58 | x86 | also, "robot voice" distortion randomly, even when a call is not bridged? |
| 18:24.13 | x86 | for example, user goes off hook, and the dialtone they hear may get distorted |
| 18:24.49 | x86 | no errors on the spans |
| 18:25.03 | x86 | callingprogress is disabled, as is busydetect |
| 18:25.32 | x86 | actually, ifconfig is showing me some framing errors, but they're only incrementing at a rate of a little under 1 per minute |
| 18:25.58 | *** join/#asterisk maurot (n=chebyte@host166.200-117-195.telecom.net.ar) |
| 18:26.09 | maurot | hi people |
| 18:26.12 | maurot | i have a question |
| 18:26.15 | x86 | any ideas? |
| 18:26.19 | maurot | i have a xp1000 board |
| 18:26.27 | maurot | anyone recommend me other board? |
| 18:26.29 | x86 | ManxPower: surely you mean x100p |
| 18:26.38 | maurot | yeap |
| 18:26.40 | maurot | xp100p |
| 18:26.41 | maurot | sory |
| 18:26.45 | x86 | err.... maurot I meant, not ManxPower |
| 18:26.50 | x86 | stupid tab completion ;) |
| 18:26.53 | maurot | jajaaj |
| 18:27.14 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 18:27.16 | x86 | maurot: Sangoma A20002d-x |
| 18:27.31 | x86 | real nice POTS line card |
| 18:27.34 | maurot | x86: and Wildcard TDM400P ? |
| 18:27.40 | b11d` | anyone know a reasonable place to list cards to sell? |
| 18:27.50 | b11d` | i've got a te411e and a sangoma a104d to sell |
| 18:27.51 | Qwell | ebay? |
| 18:27.55 | moy | coppice: so far just Thailand, but since you already supported that I suppose is not that strange :) |
| 18:28.03 | x86 | I don't know too much about digium products, except that I had all kinds of problems with their T1 cards |
| 18:28.35 | x86 | Qwell: you don't deal with analog right? |
| 18:28.40 | maurot | ok |
| 18:28.40 | Qwell | x86: I try not to |
| 18:28.51 | x86 | wish i could say the same ;) |
| 18:28.57 | x86 | I'm being forced to use analog |
| 18:29.24 | x86 | I wonder if these problems stem from irq misses or something related to the server |
| 18:29.44 | x86 | perhaps I'll call sangoma |
| 18:31.23 | korihor | moy: yes, I tested on Venezuela telco's CANTV, Movistar and Movilnet :) |
| 18:31.46 | moy | korihor: nice, thanks a lot |
| 18:32.52 | korihor | moy: you are welcome :) |
| 18:33.02 | coppice | moy: its not strange until you have to look it up in the atlas :-) |
| 18:33.33 | coppice | moy: I think the biggest surprises were Australia and New Zealand |
| 18:34.40 | moy | coppice: yeah, I have not found people from Australia nor New Zealand using it yet though ... did you find anyone to test DTMF/R2? |
| 18:35.07 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 18:35.40 | coppice | find people isn't hard. getting around to testing with them may be. actually the code has been pretty much does for ages |
| 18:36.20 | *** join/#asterisk CVirus (n=Satan@196.218.41.31) |
| 18:36.37 | coppice | I don't fully support Thailand. some people use my code there, but others needed some weird billing pulse arrangement that I never found proper info on |
| 18:38.52 | korihor | moy: i can test DTMF/R2 for outgoing calls :) |
| 18:39.06 | *** join/#asterisk pakipenguin (n=pkbill@linuxpakistan/admin/pakipenguin) |
| 18:39.40 | coppice | korihor: I know you can. maybe I should try to get you a protocol module to try :-) |
| 18:39.59 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:40.12 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 18:40.25 | moy | there you go, korihor to the rescue ... coppice code support DTMF/R2, mine does not yet, need to find the spec and see what I can do |
| 18:40.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:40.58 | coppice | if you've layered the R2 and MFC parts well, doing DTMF/R2 is trivial |
| 18:41.27 | moy | good news then ... the truth is most of my problems have been chan_zap |
| 18:41.43 | javb | IS there a way i can change the order of how Asterisk play files on musiconhold? |
| 18:43.15 | korihor | moy: I have done DTMF/R2 for Unicall, coppice has a copy that I sent him :p |
| 18:43.15 | coppice | that's one of the reasons why I made chan_unicall separate. there was actually a version with unicall support merged into chan_zap at one time. even now chan_unicall.c is still a lot like the chan_zap code I started from |
| 18:44.21 | moy | korihor: ah, I did not know that |
| 18:44.47 | coppice | moy: I think he means he found the info on what is required. he sent me that |
| 18:45.29 | coppice | and its a trivial dialing out of the destination number as DTMF, with no ANI or other info |
| 18:46.13 | moy | korihor: if that's what you meant, I would appreciate if you can send me that as well ... coppice, telcos really use DTMF/R2 or this is more like for PBX connectivity? |
| 18:46.56 | korihor | moy: I too sent patch :p |
| 18:47.02 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:47.05 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:47.29 | coppice | in venezuela, and I think one other place, they make you use separate E1s for incoming and outgoing. the incoming is provided as MFC/R2 and the outgoing as DTMF/R2 |
| 18:47.58 | moy | wtf?! |
| 18:47.58 | korihor | coppice: that's rigth |
| 18:48.02 | coppice | korihor: I don't remember that, but maybe its somewhere in my e-mail |
| 18:48.18 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 18:50.22 | coppice | moy: you had a really easy time, because you had a reference model. A huge percentage of the work in developing my R2 code was actually finding the info about the various countries |
| 18:50.44 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 18:50.58 | korihor | moy: give me your email to send you the same email that I sent to coppice |
| 18:52.30 | coppice | moy: has chan_zap changed much between 1.4 and 1.6? |
| 18:53.24 | pakipenguin | hi |
| 18:53.43 | pakipenguin | has anyone been able to get asterisk registered tu huawei softx3000? |
| 18:53.46 | [TK]D-Fender | coppice: Given 1.60 was running off 1.4 for so long, basically no. |
| 18:53.59 | moy | coppice: agreed, the reference is everything here ... regarding chan_zap, not much, at least not much in the areas I had to touch to implement R2, once I had the patch for 1.6 it was pretty simple port it for 1.4 ... porting it to 1.2 was more work though |
| 18:54.40 | coppice | [TK]D-Fender: I think you mean zaptel. I'm talking about chan_zap |
| 18:55.01 | [TK]D-Fender | coppice: Ah yes, indeed a bit separate, and No I don't actually know. |
| 18:55.34 | coppice | and now its all going dahdidahdida |
| 18:55.43 | Juggie | coppice, there is no chan_zap in 1.6 :) |
| 18:56.04 | coppice | so its chan_charlie now? |
| 18:56.15 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-018-148.mycingular.net) |
| 18:56.44 | moy | coppice: yeah, that's a pain, I will have to add some ifdefs to libopenr2 to handle dahdi vs zaptel and I have not merged yet trunk because I have a sangoma card for testing, I don't think wanpipe trunk works with dahdi yet |
| 18:57.24 | Juggie | sheesh, make digium send you a card :) |
| 18:57.29 | coppice | sangoma hardly ever works. it seems to break on every tiny kernel update :-\ |
| 18:57.52 | moy | coppice: really? well, I don't upgrade kernel that often, that's why I have not noticed |
| 18:58.07 | moy | Juggie: Mark was supposed to, but I just got the invoice, not the card :) |
| 18:58.08 | Juggie | most normal people dont :) |
| 18:58.18 | Juggie | haha |
| 18:58.23 | moy | Juggie: invoice at 0's of course |
| 18:58.28 | Juggie | well, the card will follow, mail moves faster then boxes. |
| 18:58.39 | coppice | I have no expectation of a fresh wanpipe built working, especially for 64 bit. its a pain almost every time |
| 18:58.59 | Juggie | coppice, we have 64bit wanpipe working just fine on centos |
| 18:59.11 | moy | coppice: I was lucky then, I have an Ubuntu 64 bit box with sangoma working |
| 18:59.18 | Juggie | an entire DS3 of sangoma cards even |
| 18:59.28 | Juggie | all purring along @ 64bit |
| 18:59.48 | coppice | centos has a fairly old kernel. FC9 wouldn't work with sangoma last week, when I tried the last build |
| 19:00.05 | Juggie | what does FC9 have? |
| 19:00.14 | Qwell | probably 2.6.25 |
| 19:00.14 | Juggie | centos is running 2.6.18 |
| 19:00.19 | coppice | 2.6.25.9-76.fc9 |
| 19:00.40 | [TK]D-Fender | coppice: Yeah, well * won't compile the latest stable apparently either |
| 19:00.57 | [TK]D-Fender | coppice: bleeding edge is a messy place to be :) |
| 19:01.03 | coppice | but when the sangoma drivers are working, they are far less trouble than any of the alternatives |
| 19:01.09 | Juggie | well i do hope FC9 is a dev box |
| 19:01.14 | Juggie | if it was a server i'd have to slap you |
| 19:01.29 | coppice | I have no servers. I develop |
| 19:01.47 | Juggie | none the less FC is on the 'bleeding edge' so meh :) |
| 19:02.20 | coppice | they were doing nicely with 6, 7 and 8, but 9 has gone badly wonky |
| 19:04.13 | x86 | anyone have any ideas at all on this: http://pastebin.ca/1073640 ? |
| 19:05.51 | x86 | tricky analog issue that I can't figure out |
| 19:05.56 | Juggie | i think centos has a nice desktop personally |
| 19:06.01 | x86 | no errors on the T1 interfaces or anything.... |
| 19:06.11 | Juggie | last time i tried it it did not work properly with the suspend in my laptop though |
| 19:06.42 | coppice | Linux used to work really well with laptops in 1999. far better than today |
| 19:06.50 | Juggie | now i have a new laptop though, and i cant get 3d to work properly. |
| 19:07.11 | x86 | so no on-topic discussion today? |
| 19:07.18 | Juggie | well i cant get anything other then the vesa driver to work as i recall |
| 19:08.00 | Juggie | x86, ask sangoma. |
| 19:08.16 | x86 | I did |
| 19:08.28 | x86 | but they're taking thier time getting back to me |
| 19:08.34 | pakipenguin | x86, hey!! |
| 19:08.41 | x86 | pakipenguin: hey man, long time no see |
| 19:08.48 | x86 | pakipenguin: you do call center VoIP right? |
| 19:08.56 | pakipenguin | not anymore |
| 19:09.03 | x86 | know anything about analog? |
| 19:09.13 | pakipenguin | right now i am pulling my hair due to a bloody huawei softx3000 switch |
| 19:09.14 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 19:09.14 | pakipenguin | :( |
| 19:09.22 | emiller | im getting callback on my phones. i have asterisk installed in my Clinic A location. If I (say, Clinic B) calls Clinic A, call connects fine. But when Clinic A hangs up first, then clinic B hangs up second, Clinic A receives a callback. Anywhere in the config i can look around? |
| 19:09.55 | x86 | emiller: I get the same problem from time to time... if you find out how to fix it, please let me know |
| 19:10.05 | x86 | emiller: using analog stations on both ends? |
| 19:10.10 | emiller | you got it |
| 19:10.16 | x86 | and when you answer the callback, you just get dialtone right? |
| 19:10.20 | emiller | yup |
| 19:10.36 | x86 | yeah same issue as one of the many I'm facing right now |
| 19:10.52 | pakipenguin | hmm |
| 19:11.00 | emiller | wierd. |
| 19:11.14 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 19:11.35 | lesouvage | x86 emiller: do you both use callfiles. If you don't have a proper hard return at the last line of a callfile some strange behaviour might uccur. |
| 19:11.52 | rwaite | can someone please explain ResetCDR() to me? i am confused on the v option, what variables is it talking about? |
| 19:11.54 | x86 | not using a call file |
| 19:12.06 | emiller | no, i don't lesouvage. thank you for the insight, though. |
| 19:12.19 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-018-148.mycingular.net) |
| 19:13.22 | lesouvage | x86 emiller: what does the cli show when it happens. |
| 19:14.45 | x86 | not much |
| 19:14.52 | x86 | normal call clearing |
| 19:14.58 | x86 | doesn't show the ringback |
| 19:16.43 | emiller | <PROTECTED> |
| 19:16.43 | emiller | <PROTECTED> |
| 19:16.43 | emiller | <PROTECTED> |
| 19:16.43 | emiller | [Jul 16 15:16:05] WARNING[9128]: chan_zap.c:6538 ss_thread: CallerID returned with error on channel 'Zap/2-1' |
| 19:16.44 | emiller | <PROTECTED> |
| 19:16.53 | emiller | sorry for posting the code, pastebin is getting blocked for me |
| 19:17.00 | emiller | lesouvage ^^ |
| 19:17.37 | Strom_C | emiller: there's more than one pastebin, you know |
| 19:17.39 | Strom_C | ~pb |
| 19:17.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:17.51 | emiller | its getting websensed. |
| 19:18.03 | Strom_C | all of them? |
| 19:18.06 | emiller | trying... |
| 19:18.17 | x86 | why would a pastebin get filtered out via websense? |
| 19:18.22 | x86 | Strom_C: did you look at my paste? |
| 19:18.28 | x86 | http://pastebin.ca/1073640 |
| 19:18.30 | Strom_C | no |
| 19:18.38 | x86 | what do you take of that? |
| 19:19.15 | Kobaz | shit |
| 19:19.20 | Kobaz | asterisk just took a dump |
| 19:19.21 | Strom_C | looks like a bunch of text |
| 19:19.29 | x86 | Strom_C: heh... indeed it is |
| 19:19.43 | Kobaz | every so often asterisk starts chewing up 100% cpu and can't take any more calls, and completely is unusable |
| 19:19.56 | x86 | Kobaz: what version? |
| 19:19.59 | emiller | thank you Strom_C, one of the other pinks worked. |
| 19:20.03 | emiller | links* |
| 19:20.06 | emiller | http://paste.lisp.org/display/63783 |
| 19:22.19 | Kobaz | x86: 1.4.14 |
| 19:22.32 | Kobaz | we haven't moved to anything newer yet since we're still testing |
| 19:22.59 | x86 | perhaps you should try an upgrade and see if your problems persist |
| 19:24.29 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-018-148.mycingular.net) |
| 19:25.19 | Kobaz | this doesnt happen to often |
| 19:25.27 | Kobaz | we should push the upgrade through soon |
| 19:33.36 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:34.58 | lesouvage | x86: I think line 166 and 167 might not be ok. I check a server with a sangoma card. |
| 19:36.28 | x86 | pakipenguin: jump on ssh :) |
| 19:36.34 | x86 | lesouvage: on my config files? which one? |
| 19:37.02 | *** join/#asterisk riddlebox (n=me@75-128-170-26.static.stls.mo.charter.com) |
| 19:39.26 | lesouvage | x86: your wanpipe file in your paste http://pastebin.ca/1073640 |
| 19:43.37 | *** join/#asterisk bmg505 (n=leon@196-209-77-195-tbnb-esr-2.dynamic.isadsl.co.za) |
| 19:45.11 | lesouvage | x86: I have te_ref_clock=1 where TE_clock=master |
| 19:46.39 | lesouvage | x86: where you ahve te_ref_clock=0 . I'm not really an expert on this field but it has to the with the timer and where it gets its timing from. AFAIK with this setting the provider takes care of the timing and the asterisk box is "following". |
| 19:47.29 | lesouvage | x86: you could give it a try |
| 19:48.52 | riddlebox | has anyone gotten 3com phones working with asterisk |
| 19:51.13 | *** join/#asterisk CVirus (n=Satan@196.218.41.31) |
| 19:51.13 | x86 | lesouvage: that's the reference clock, if you're getting timing from an external source and want to use it as a reference |
| 19:51.23 | CVirus | I wonder what is wrong with this dialplan http://rafb.net/p/1ZJTbN13.html |
| 19:51.38 | x86 | lesouvage: in this case, we're not getting timing anywhere on this card, only providing timing, so we don't want a reference clock |
| 19:52.36 | CVirus | the false destination is always executed |
| 19:52.45 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 19:55.29 | lesouvage | x86: but why? How do you synchronize with the telco. I'm asking because I had issues too and it was solved with other settings for timing. |
| 19:58.42 | [TK]D-Fender | CVirus: And your pastebin of the CALL pleas.... |
| 20:00.27 | [TK]D-Fender | lesouvage: "master" = * SETS timing. "normal' = TAKES timing. |
| 20:01.42 | CVirus | [TK]D-Fender: http://rafb.net/p/D8IN5t33.html |
| 20:02.25 | CVirus | [TK]D-Fender: want me to paste the whole context too ? |
| 20:02.55 | [TK]D-Fender | CVirus: no need. It is "Zap/1" and it does the Goto just fine as the "true" |
| 20:03.09 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 20:03.13 | x86 | lesouvage: channel banks don't need to sync to the telco, and in fact, you can't sync across cards anyway (telco is on another T1 card) |
| 20:03.19 | [TK]D-Fender | CVirus: HOWEVER.... |
| 20:03.23 | [TK]D-Fender | CVirus: :) |
| 20:03.27 | x86 | anyone know the default IP of an Adit 600? |
| 20:03.53 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 20:04.02 | CVirus | [TK]D-Fender: the second condition is executed too ? |
| 20:04.03 | [TK]D-Fender | CVirus: Like a schmuck you just let it roll on over the code that FALSE leads you to thus overriding your intent. |
| 20:04.12 | CVirus | i see |
| 20:04.16 | CVirus | i'll have to add a Goto then |
| 20:04.21 | [TK]D-Fender | CVirus: SMRT |
| 20:04.43 | [TK]D-Fender | CVirus: not necessarily, but thats one option. |
| 20:05.04 | CVirus | what's the other ? |
| 20:05.11 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 20:07.41 | [TK]D-Fender | CVirus: http://pastebin.com/m32718685 |
| 20:07.41 | lesouvage | [TK]D-Fender: and what is the te_ref_clock=0 setting? |
| 20:07.55 | [TK]D-Fender | lesouvage: I *think* 0 = normal. |
| 20:08.43 | CVirus | [TK]D-Fender: thanks |
| 20:09.15 | [TK]D-Fender | CVirus: My way works with a minimum of Goto's, but is less readable. |
| 20:09.33 | [TK]D-Fender | CVirus: I would likely do it the other way personally. |
| 20:10.04 | CVirus | [TK]D-Fender: thanks alot |
| 20:10.59 | lesouvage | [TK]D-FEnder: I thought that te_ref_clock=1 is that the master in the system gets it timing from the telco (again, I'm not isdn expert, just learning by making mistakes so perhaps this is clear nonsense) |
| 20:13.14 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 20:13.40 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 20:13.58 | Defraz | . |
| 20:15.32 | lesouvage | x86: this comes from the sangoma site TE_REF_CLOCK=1 #use the incoming clock from port1 as the MASTER clock I think it has to use an incoming clock from somewhere to be synchronized with the outside world. I don't think the outside world will synchronize with the Asterisk box ;-) |
| 20:15.39 | *** join/#asterisk anthm (n=anthm@mbe0736d0.tmodns.net) |
| 20:17.47 | freezey | i setup some phones on an adhoc instead of fixed and i set *11 to act as my logon feature.. but each time when i dial it its telling me the number is not inservice.. i am wonderng why its not treating this as a featurecode |
| 20:17.58 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:18.35 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 20:18.37 | [TK]D-Fender | freezey: What miracle "featurecode" are you talking about? |
| 20:18.55 | [TK]D-Fender | freezey: pastebin a failed attempt with SIP debug and include your dilaplan. |
| 20:19.20 | tzanger | gplpv tht is what it was called |
| 20:19.36 | *** part/#asterisk timothy_jones (n=root@207.90.4.3) |
| 20:20.02 | freezey | [TK]D-Fender: http://www.pastebin.ca/1073708 |
| 20:20.52 | javb | im converting mp3 to gsm ... but losing too much quality |
| 20:21.06 | x86 | lesouvage: i'm telling you... you can't sync between different cards |
| 20:21.13 | x86 | sangoma doesn't support it |
| 20:21.30 | javb | <PROTECTED> |
| 20:21.44 | [TK]D-Fender | freezey: Executing [*@from-sip-external:1] NoOp("SIP/192.168.1.200-09b9c998", "Received incoming SIP connection from unknown peer to *") in new stack <- Asterisk only got the "*". Go configure your phone. |
| 20:21.55 | pputman- | That's an awesome term. Featurecode. |
| 20:22.17 | [TK]D-Fender | freezey: And I can't imagine how this is going to work, allowing UNAUTHED calls into your system. You will have no way to know what device is calling. |
| 20:22.24 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 20:22.40 | [TK]D-Fender | freezey: Your approach is very broken. Stop now before you hurt yourself. |
| 20:23.00 | lesouvage | x86: OK |
| 20:23.10 | freezey | [TK]D-Fender: its worth seeing if its possible |
| 20:23.11 | [TK]D-Fender | freezey: And you seem to be running off a GUI. Another dead-end. Never quote some scripted prompt (like you just did) as if it meant anything. |
| 20:24.10 | ManxPower | [TK]D-Fender: freezey just seems like he's deliberately trying to do things the hard way, doesn't he? |
| 20:24.23 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 20:24.42 | [TK]D-Fender | ManxPower: Wants creative, clue[-1] |
| 20:24.57 | ManxPower | There is no place for creativity in telecom 8-) |
| 20:25.37 | [TK]D-Fender | ManxPower: Speaking like a true burnout :) |
| 20:25.50 | [TK]D-Fender | fetches some marshmallows |
| 20:25.57 | ManxPower | being creative is what burns you out |
| 20:27.37 | [TK]D-Fender | alrighty, heading home. BBIAB |
| 20:28.01 | lanning | entropy is the only true winner :) |
| 20:28.21 | x86 | anyone know how to see what a T1 interface's timing is set to on an Adit 600? |
| 20:28.28 | x86 | pakipenguin: you leave me? |
| 20:28.45 | ManxPower | x86: it would default to "receive sync from network", but I don't know how to see if it's been changed. |
| 20:29.16 | x86 | ManxPower: that's what I thought... sangoma said it has to be running as master.... |
| 20:29.17 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 20:29.31 | x86 | everyone keeps telling me that the problems are timing related, but I can't see how.... |
| 20:30.10 | ManxPower | x86: sangoma lied |
| 20:30.39 | x86 | that's what I thought! |
| 20:30.46 | ManxPower | you can't connect to a T-1 telco connection if you are a sync master |
| 20:30.59 | x86 | every channel bank I've ever touched defaults to slave timing |
| 20:31.21 | x86 | and asterisk was setup for master timing |
| 20:31.29 | x86 | still is |
| 20:31.39 | x86 | they told me to try changing asterisk's timing to normal |
| 20:31.50 | x86 | "because the channel banks must be in master mode" they said |
| 20:32.46 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 20:34.39 | *** join/#asterisk _khan (n=shariq@116.71.11.125) |
| 20:36.28 | justdave | would be nice if Polycom's dialplan stuff would let you specify a separate dialplan for each line (so the pattern it followed would depend on which line you had selected) |
| 20:36.34 | justdave | or is there some way to do that and I'm missing it? :) |
| 20:36.59 | x86 | justdave: think the dialplan is global on those things |
| 20:37.13 | justdave | IP501 connected to two different asterisk servers, which have different dialplans serverside, would be nice to have that reflected in the phone, too |
| 20:37.38 | justdave | (one's my home phone system, the other is work's system :) |
| 20:38.14 | *** join/#asterisk bl4q (i=Bl@dslb-088-067-045-022.pools.arcor-ip.net) |
| 20:38.48 | x86 | you're beyond me... I could never get one to associate properly with two servers at the same time |
| 20:39.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:41.14 | justdave | each line button has it's own server settings |
| 20:41.16 | tzanger | justdave: I don't have the phone do anything smart |
| 20:41.24 | tzanger | no dialplan in the phone, let each server decide |
| 20:41.39 | beek | ss |
| 20:41.44 | justdave | the best dialplan for a polycom: ".T" :) |
| 20:42.00 | justdave | I suppose "x.T" actually |
| 20:44.42 | freezey | ManxPower: so your saying setting up the phones ad-hoc is a bad idea? |
| 20:45.02 | *** join/#asterisk Hydrant (n=aj@74.210.126.107) |
| 20:45.28 | x86 | Hydrant: how's the openmoko? |
| 20:45.36 | Hydrant | hey all... how much data transfer is really required by SIP? I'm looking at data plans for my openmoko cell phone... |
| 20:45.44 | Hydrant | x86: works good |
| 20:45.55 | Hydrant | x86: for an experimental device :-) |
| 20:47.35 | mandd | channel of type 'SIP' (cause 58 - Bearer capability not available) |
| 20:47.47 | x86 | Hydrant: SIP itself uses very little... but I'm assuming you wanted an estimate with RTP included? :) |
| 20:47.59 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 20:48.01 | jm|laptop | heh |
| 20:48.05 | Hydrant | RTP ? |
| 20:48.11 | nny_1 | anyone know of a nice php script to generate a "company directory" from sip.conf? |
| 20:48.13 | x86 | Hydrant: depends on codec... usually you can do G711u at around 80-100kbps |
| 20:48.24 | nny_1 | something where a user could look and see what extension someone was |
| 20:48.29 | x86 | nny_1: nice and php don't work in the same sentance |
| 20:48.30 | nny_1 | is* |
| 20:48.32 | nny_1 | ha |
| 20:48.50 | Qwell | nny_1: awk |
| 20:48.50 | x86 | Hydrant: real time protocol, the protocol that the voice transports over after SIP initiates the session |
| 20:48.52 | Hydrant | no I have to always be connected to receive a call via SIP? |
| 20:49.03 | x86 | yes |
| 20:49.15 | Hydrant | hrrm... perhaps I can get around that somehow ? |
| 20:49.43 | Hydrant | like have an intermediate box accept do something to activate the network on my phone |
| 20:50.01 | Hydrant | I wonder how push email works... |
| 20:50.03 | nny_1 | eh nm |
| 20:50.12 | nny_1 | easier yet is just to let the users dial the company directory |
| 20:50.17 | nny_1 | >< |
| 20:51.45 | _khan | any website where php agi scripts present |
| 20:52.20 | justdave | Hydrant: push email requires the support of the phone company |
| 20:52.38 | Hydrant | hrrm... so how can I do push sip |
| 20:52.43 | justdave | your best bet on emulating it (if you have control over the phone) is for your intermediate server to send an SMS telling you to connect |
| 20:52.51 | Hydrant | I was kinda thinking that |
| 20:53.34 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:53.47 | Hydrant | so use an intermediate asterisk box to handle the call, delay a bit while it SMS' me... then if I connect in the timeout time, forward the call via SIP otherwise take a voicemail |
| 20:53.49 | Hydrant | hows't that sound |
| 20:54.28 | mandd | bah |
| 20:54.30 | mandd | fax is working |
| 20:54.32 | mandd | muhaha |
| 20:54.40 | mandd | great success |
| 20:55.41 | mandd | thanks again [TK]D-Fender, ManxPower & coppice |
| 20:56.58 | *** join/#asterisk mmarcost (n=mmarcos@151.193.220.29) |
| 20:57.41 | _khan | can some one recommend me, where can i download php agi scripts?? |
| 20:57.49 | x86 | dude |
| 20:57.51 | x86 | DONT USE PHP |
| 20:57.53 | x86 | gah |
| 20:58.02 | _khan | why |
| 20:58.03 | Nugget | seconds that advice |
| 20:58.46 | Nugget | php is a marginally-useful (but flawed) html markup language. trying to treat it like a general-purpose scripting language or development language is a terrible idea. |
| 20:59.00 | x86 | yeah php is horrible |
| 20:59.11 | javb | when i have background playing via a wired phone, works great, but when calling from my cellphone, background sometimes will listen to me and sometimes no ! |
| 20:59.13 | javb | any idea? |
| 20:59.35 | mmarcost | has somebody installed spandsp/fax addon on 1.4 successfully? |
| 20:59.49 | [TK]D-Fender | _khan: www.google.com |
| 21:00.09 | [TK]D-Fender | mmarcost: tons of people. |
| 21:00.16 | _khan | thanks |
| 21:00.45 | javb | ...? |
| 21:00.52 | tzanger | damn, even with gplpv it still hits 100% cpu |
| 21:01.35 | mmarcost | [TK]D-Fender: do you know if you must recompile asterisk after installing the fax addons? |
| 21:02.19 | [TK]D-Fender | mmarcost: No. |
| 21:02.45 | Hydrant | is this calc right? Assuming 100kbps for transfer... that's 100,000kbits / s, or 12KB/s ? |
| 21:03.07 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 21:03.16 | Strom_C | Hydrant: way wrong |
| 21:03.22 | javb | this is the context where i have my calls comming http://pastebin.com/m368913c .... why is it working with normal phones not with cellphones ? ([TK]D-Fender ? any idea?) |
| 21:03.27 | Strom_C | to convert kilobits into kilobytes, you divide by eight. |
| 21:03.33 | Hydrant | right |
| 21:03.40 | Hydrant | then by 1024bytes / kb |
| 21:03.49 | Strom_C | no. |
| 21:04.03 | Strom_C | you said one hundred thousand kilobits, not one hundred thousand bits |
| 21:04.34 | Hydrant | k... 100,000bits |
| 21:04.52 | mmarcost | [TK]D-Fender: that's what I believed, since my * is crashing when rx, I'm not sure if it's related to * version or that I'm using E1 ISDN card instead of FXO |
| 21:05.18 | mmarcost | don't know if T1/E1 cause much problem with fax |
| 21:05.41 | Hydrant | but that's still 12 kilobytles / second right ? |
| 21:06.51 | _khan | one side voice in Recordings is very low (called party voice), any idea?? |
| 21:07.04 | [TK]D-Fender | javb: What the hell is that dialplan snippet supposed to show me? |
| 21:07.14 | [TK]D-Fender | javb: where the failed call with SIP debug? |
| 21:07.56 | [TK]D-Fender | mmarcost: its a question of matching the right app & spandsp versions. |
| 21:08.41 | xiando | [TK]D-Fender: hours later and you're still here, helping out. Impressive & nice. Thank you for doing this. |
| 21:09.00 | _khan | [TK]D-Fender: How can i increase the volume level of called number voice in Recordings?? |
| 21:09.07 | _khan | in Zap channels |
| 21:09.17 | [TK]D-Fender | _khan: what "recordings"? |
| 21:09.20 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:09.20 | javb | [TK]D-Fender, complete dial plan http://pastebin.com/m43e6f5c5 (problem: autoattendant, works great from a normal phone, background CANT hear my types when using cellphone |
| 21:09.49 | _khan | when i record the call using Monitor |
| 21:09.52 | [TK]D-Fender | javb: Again another wste... wheres the FAILED CALL? |
| 21:09.58 | [TK]D-Fender | waste* |
| 21:10.29 | [TK]D-Fender | _khan: Go play with your rxgain & txgain in zapata.conf |
| 21:10.39 | ManxPower | javb: I suspect just increasing your rxgain by a couple of db will fix it |
| 21:11.03 | _khan | [TK]D-Fender: what i need to increase rx or tx?? |
| 21:11.27 | [TK]D-Fender | _khan: ManxPower has already clarified that for you |
| 21:11.31 | _khan | and what is the normail values or rxgain and txgain |
| 21:11.40 | [TK]D-Fender | _khan: 0 |
| 21:11.55 | ManxPower | [TK]D-Fender: no, my response was for the no-cell-ivr |
| 21:12.21 | [TK]D-Fender | ManxPower: answers both actually, not by intent ;) |
| 21:12.34 | _khan | [TK]D-Fender: then what is your response? |
| 21:12.46 | [TK]D-Fender | _khan: Same. rxgain. |
| 21:12.55 | *** part/#asterisk Hydrant (n=aj@74.210.126.107) |
| 21:12.56 | _khan | [TK]D-Fender: Thanks :) |
| 21:14.06 | *** join/#asterisk CVirus (n=Satan@82.201.174.44) |
| 21:14.07 | javb | [TK]D-Fender, there is no SIP DEBUG info, because hence background is not listening me, there is no SIP signalling activity ... just a call coming from a T1, backgrounding a file, and me, with cell phone, pressing any of those options available in that context. |
| 21:14.13 | _khan | [TK]D-Fender: my rxgain is already on 2, can i increase it more?? |
| 21:14.28 | javb | ManxPower, reload chan_zap.so will reload that? |
| 21:14.35 | mmarcost | [TK]D-Fender: that was my fear, I suffered a lot configuring R2/Unicall |
| 21:16.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:17.34 | _khan | [TK]D-Fender: my rxgain is already on 2, can i increase it more?? |
| 21:17.34 | [TK]D-Fender | javb: max debug would show DTMF detection. |
| 21:18.01 | [TK]D-Fender | _khan: Yes, and note that you will have to reload chan_zap or restart * completely to take effect |
| 21:18.21 | javb | [TK]D-Fender, how can i turn sip debug off? |
| 21:18.36 | Qwell | javb: just like that |
| 21:18.47 | xiando | sip set debug off |
| 21:20.02 | javb | [TK]D-Fender, how can i set MAX DEBUG? |
| 21:22.55 | [TK]D-Fender | javb: .... "core set debug 10" |
| 21:24.55 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 21:25.16 | javb | [TK]D-Fender, that cmd doesnt exist... did this http://pastebin.com/m1bc1f27c |
| 21:25.55 | freezey | [TK]D-Fender: got it to work so the user can login via extension number and private password.. only problem is is that no matter what i still had to set a static device number to the phone so in the end its the same thing as before still setting something statically to each phone although anybody can login to the phone its still known by some sort of a number |
| 21:26.19 | x86 | javb: core set verbose 10 |
| 21:27.24 | [TK]D-Fender | javb: That'll do. |
| 21:27.42 | [TK]D-Fender | freezey: And that description says very little. PASTEBIN |
| 21:28.06 | javb | Verbosity was 100 and is now 2147483647 ---- Core debug is at least 2147483647 ......... |
| 21:28.20 | x86 | _ShrikE: hey you around? |
| 21:28.31 | javb | and, im still getting the same as before . . .. |
| 21:28.39 | javb | no MORE verbose that before ! ! |
| 21:28.48 | javb | this is so weird, first time im having this issue |
| 21:32.10 | _ShrikE | x86: yes |
| 21:34.33 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:35.54 | javb | Any idea guys ? |
| 21:36.13 | freezey | [TK]D-Fender: http://pastebin.ca/1073775 i got it so the user can now logon BUT the phones have the number like 5001 set to them but when the phone is up you can press *11 and go through the login prompt and it logs you in... and your using your extension(202) now and its mobile you can login to anyphone.. but still in the end i still have to set that 5001 to the phone |
| 21:36.16 | *** join/#asterisk eclark (n=eclark@75-164-254-175.ptld.qwest.net) |
| 21:36.49 | eclark | i'm having a sip problem i'm wondering if I can get some help on: |
| 21:37.22 | x86 | _ShrikE: when you set an Adit 600 T1 controller up as "internal" timing, that means provide timing or receive timing? |
| 21:37.26 | eclark | i can dial out, but i get an error from my sip provider unless i've got their softphone running - in which case, it rings the softphone, then rings the dialed number, and connects the two |
| 21:37.28 | x86 | provide, right? |
| 21:37.53 | x86 | eclark: the error would be helpful maybe? |
| 21:38.30 | eclark | well, it doesn't really seem to give any errors |
| 21:38.45 | eclark | is there no way to set it up to have a one-way connection? |
| 21:39.06 | eclark | like, I want this to just send outbound calls with a sound file for server monitoring |
| 21:39.40 | eclark | i'm sure this has something to do with how i've got my extensions.conf set up, but i haven't a clue where to start looking to fix it |
| 21:39.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:40.02 | *** join/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 21:40.14 | Nuitari | Hi ? |
| 21:40.24 | unpaidbill | nope |
| 21:40.33 | Nuitari | I've got a weird problem with making a call using carrierx.us to a peer using les.net |
| 21:41.05 | Nuitari | it seems that les.net is using the caller id received from carrierx as the username, instead of the username of les.net, is there any way to tell asterisk to ignore the incoming username on a peer? |
| 21:41.39 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 21:42.26 | WhiteWolf | Nuitari: that's how it matches which entry to use... except on the guest entry |
| 21:42.27 | tompaw | Yeah, my asterisk handled 32 concurrent g729 connections :-) |
| 21:42.46 | tompaw | In fact, CPU load didn't exceed 0.07 :P |
| 21:44.47 | Nuitari | which implementation? |
| 21:45.02 | javb | And better yet, it detect from the cell phone the " * " but not the numbers ! ! ! |
| 21:45.51 | _ShrikE | x86: yes, it uses itself for timing |
| 21:51.17 | _khan | how much g729 take actual bandwidth? |
| 21:54.19 | mvanbaak | hh/18 |
| 21:54.54 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 21:55.47 | WhiteWolf | _khan: 6.4-11.4kbit depending on implementation |
| 21:56.06 | WhiteWolf | standard is 8 |
| 21:56.32 | _khan | WhiteWolf: implementation means??? |
| 21:56.38 | _ShrikE | _khan: with all the included over head it gets closer to 24 |
| 21:57.21 | WhiteWolf | _khan: G729 defines 8kbit/s (one way, of course) --- but there are 6.4kbit/s and 11.4kbit/s variants |
| 21:57.38 | _khan | _ShrikE: WhiteWolf says max upto 11.4, which one i treat it as correct |
| 21:58.09 | WhiteWolf | _khan: assuming the highest, two-way comms would be about 22.8, with overhead 24 |
| 21:58.36 | _khan | thanks to both WhiteWolf & _ShrikE, now i understand. |
| 21:58.42 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 22:01.35 | [TK]D-Fender | freezey: Still around? |
| 22:03.08 | eclark | okay, i think i figured out my problem, but now I'm getting NOTICE: Failed to authenticate on INVITE to '"asterisk" |
| 22:03.37 | *** join/#asterisk chinku (n=chinku@cpe-72-225-237-124.nyc.res.rr.com) |
| 22:03.42 | [TK]D-Fender | freezey: FreePBX is NOT supported in here and what you are attempting is completely outside of any reasonable expectation to implement given that it will wipe your configs the next time you commit a change. |
| 22:04.20 | *** join/#asterisk kimc (n=kimc@c-71-205-244-134.hsd1.mi.comcast.net) |
| 22:04.54 | kimc | greetings asterisk |
| 22:05.31 | kimc | with the console running iax2 debug |
| 22:05.35 | [TK]D-Fender | freezey: FreePBX is a fresh hot steaming pile of platypus manue. The fumes alone are classes as a WMD, and our ability to help you has been equally destroyed. You are in the WRONG channel. |
| 22:05.46 | [TK]D-Fender | manure* |
| 22:06.03 | jameswf-home | ~freepbx |
| 22:06.04 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:06.08 | WhiteWolf | d-mn... nice roll :P |
| 22:06.14 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:06.21 | jameswf-home | rick roll? |
| 22:06.25 | kimc | the time in the DATE TIME field is ~2hours behind system time -? |
| 22:06.59 | jameswf-home | kimc: sounds like a timezone issue |
| 22:07.04 | kimc | right |
| 22:07.21 | kimc | the date command returns local time |
| 22:07.23 | WhiteWolf | is it exactly 2 hours behind, or approx 2 hours behind |
| 22:07.31 | kimc | lemme check |
| 22:08.18 | kimc | waiting for an iax reg to expire.. |
| 22:10.43 | kimc | its 2 hours 26 secs behind |
| 22:12.11 | WhiteWolf | the fact it's not exactly 2 hours behind indicates something very silly |
| 22:12.18 | WhiteWolf | perhaps the SQL server time is off? |
| 22:12.32 | kimc | yes.. no SQL here |
| 22:12.44 | WhiteWolf | what's storing the data you're reading |
| 22:12.58 | kimc | i'm just monitoring the console output |
| 22:13.01 | WhiteWolf | ah |
| 22:13.27 | kimc | hmm.. its not the time coming from the other end is it? |
| 22:13.35 | WhiteWolf | it might be |
| 22:13.55 | kimc | teliax is 2 hours behind here I think |
| 22:14.03 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 22:14.57 | kimc | theres a Timestamp field here |
| 22:15.10 | kimc | just printed 9 ms |
| 22:15.18 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:16.14 | WhiteWolf | i'd have to look closer at the proto to know if the time source is the other end on that reply |
| 22:16.49 | kimc | jeez.. it just printed 16:16:06 when the system time was 18:15:47 |
| 22:17.36 | kimc | ok on looking at the proto |
| 22:17.53 | *** join/#asterisk nGbzxc (n=root@200.61.172.61) |
| 22:18.05 | nGbzxc | hi guys... some one speack in spanish? |
| 22:18.33 | Qwell | nGbzxc: I'm guessing you do. |
| 22:18.54 | nGbzxc | i've problems with rx fax... |
| 22:18.57 | WhiteWolf | kimc: if it's not causing any adverse problems, can probably ignore it |
| 22:19.24 | kimc | ok very good |
| 22:19.27 | nGbzxc | when some fax are recived, the tif image is write with lines |
| 22:19.50 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:19.53 | WhiteWolf | nGbzxc: probably sending end scanning error |
| 22:20.17 | WhiteWolf | or very poor transmission |
| 22:20.40 | nGbzxc | it problem ocurs when i send a fax on PSTN... if i send it over Internet the fax is good |
| 22:20.58 | Qwell | damn, where's coppice |
| 22:21.02 | Qwell | he'd enjoy that |
| 22:22.13 | nGbzxc | some one know some metod to debug it? |
| 22:22.49 | nGbzxc | (sorry for my english... i know that is very bad :( |
| 22:25.03 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 22:25.15 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:26.16 | nGbzxc | some body can help me ? |
| 22:29.05 | *** part/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 22:29.11 | nGbzxc | !clear |
| 22:31.46 | nGbzxc | *part |
| 22:31.48 | *** part/#asterisk nGbzxc (n=root@200.61.172.61) |
| 22:32.04 | freezey | [TK]D-Fender: i was just trying it out on that i have asterisk without freepbx running as well... i just wanted to see if it offered something tha ti was looking for |
| 22:32.24 | freezey | [TK]D-Fender: i have been reading through all those dialplan applications and functions and am still looking through them |
| 22:32.40 | freezey | [TK]D-Fender: i would much rather do this without putting freepbx on top of anything cause i think its trash as well |
| 22:34.09 | freezey | [TK]D-Fender: you still there? |
| 22:51.44 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 22:52.54 | wwalker | When I call in from the PSTN calls go through and ring the right extension. but the caller hears no ring indication. any ideas where I should look? |
| 22:52.59 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:53.37 | Qwell | are you answering, or doing any file playback or anything? |
| 22:53.45 | Qwell | before sending the call to the next destination |
| 22:55.26 | wwalker | Qwell the call goes to an AGI (adhearsion) which then dials the actual extension. DOH, AGI is autoanswering the call? |
| 22:55.38 | Qwell | probably |
| 22:56.12 | wwalker | so in an AGI how what's the proper way to fake ring? Background? or something else? |
| 22:56.35 | wwalker | (or in a dialplan that has for some odd reason already answered |
| 23:06.02 | *** join/#asterisk macros73_ (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 23:09.25 | ManxPower | wwalker: how about not auto answering? |
| 23:09.58 | ManxPower | background auto answers, as so many other things, check the app docs |
| 23:11.16 | *** join/#asterisk blq (i=Bl@dslb-088-064-157-187.pools.arcor-ip.net) |
| 23:12.35 | *** join/#asterisk TrentCreek (n=TrentCre@red1.cs.panam.edu) |
| 23:14.33 | wwalker | ManxPower: I have to transfer control to the AGI, which will probably call an extension. I need to provide a ring indication to the user during that time |
| 23:15.14 | *** join/#asterisk tristezo2k (n=seba@host158.190-224-187.telecom.net.ar) |
| 23:15.39 | tristezo2k | Hello everybody... |
| 23:16.22 | *** part/#asterisk _khan (n=shariq@116.71.11.125) |
| 23:16.29 | tristezo2k | I have an H323 related question... |
| 23:16.30 | ManxPower | wwalker: That should work just fine. find out why it isn't by looking at every app that is run in the call process and fine the one(s) that are answering when you don't want them to. |
| 23:16.43 | tristezo2k | I made an entry in the h323.conf file |
| 23:17.01 | tristezo2k | that goes to the context xxxxx on the extensions.conf |
| 23:17.33 | tristezo2k | it is a peer , context xxxxx host=a.b.c.d |
| 23:18.00 | tristezo2k | when a call comes in, instead of going to the xxxxx context it is received in the default context. |
| 23:18.08 | wwalker | ManxPower: I'm pretty sure that AGI answers immediately, you don't get a choice. |
| 23:18.15 | tristezo2k | how can I make it work? |
| 23:19.22 | wwalker | hmmm, guess not. |
| 23:20.59 | ManxPower | wwalker: I'm pretty sure it doesn't |
| 23:21.41 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 23:22.32 | wwalker | anyone have a recording of ring indication? (US) Asterisk indication module should be playing a ring indication during dialing but isn't, so I need to play one via background |
| 23:23.00 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
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| 23:43.22 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 23:55.52 | macros73_ | How do I transfer a call from an analog phone attached to an ATA that's peering with an Asterisk server? |
| 23:56.20 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 23:56.36 | jblack | macros73_: Usually with the flash button on the phone. |
| 23:56.45 | jblack | If you don't have a flash button, try tapping the hook. |
| 23:57.17 | macros73_ | jblack: Tapping the hook gets me on hold, but the typical ## doesn't initiate a blind transfer. |
| 23:57.27 | *** join/#asterisk timothy_jones (n=admin@207.90.4.3) |