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| 02:27.19 | maisonhotline | is this the proper place to seek help troubleshooting an issue? |
| 02:28.56 | maisonhotline | having an issue setting up a machine, it responds appropriately to AT&T and T-Mobile phones by playing our test audio, but with Verizon it connects and then is silent |
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| 02:56.04 | paulc | maisonhotline: what kind of connection are you bringing the call into Asterisk on? |
| 03:01.20 | maisonhotline | paulc: the server is hosted on a AT&T u-verse DSL connection |
| 03:03.19 | maisonhotline | (hardwired via lan) |
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| 04:01.24 | paulc | maisonhotline: so you're using an ITSP to bring the calls in via SIP? |
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| 04:52.06 | Milos | Could someone help me figure out what I need to do to resolve this [Nov 6 17:51:11] WARNING[7832][C-0000009b]: channel.c:5125 ast_write: Codec mismatch on channel Local/104@outgoing-0000000e;1 setting write format to g722 from ulaw native formats (ulaw) |
| 04:52.13 | Milos | the calls still go through so I'm not sure what's happening |
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| 05:12.36 | Milos | also I have a problem with dtmfmode=rfc2833 skipping some tones when they're typed quickly |
| 05:12.41 | Milos | where is a setting I can use to tune this? |
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| 06:44.05 | UncleKiwi | hey people, I am a user of asterisk and I am getting lots out of it! I need to learn all about it in the fastest way what do you suggest are there any good documentation ( that will not put me to sleep) or video anyone can recommend ? |
| 06:45.53 | UncleKiwi | I want to understand they big picture not just get it working |
| 06:46.01 | UncleKiwi | I need to make it secure |
| 06:46.47 | ChannelZ | ~book |
| 06:46.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 06:46.58 | UncleKiwi | thank you ChannelZ |
| 06:57.08 | ChannelZ | Sure |
| 06:58.21 | UncleKiwi | I have ordered it in paaperback and my kindel is charging |
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| 07:03.30 | UncleKiwi | hey ChannelZ I have an FXO device SPA3102 tomorrow I need to configure it to take calls inbound. Does this require much cpu ? |
| 07:03.43 | UncleKiwi | on the asterisk server it's self |
| 07:03.51 | UncleKiwi | ? |
| 07:05.20 | UncleKiwi | inbound into asterisk |
| 07:05.27 | UncleKiwi | from the pstn |
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| 07:10.15 | ChannelZ | No not really |
| 07:12.40 | UncleKiwi | is it hard to have two asterisk boxes running in a clustered way ? |
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| 07:13.33 | UncleKiwi | ie two physical asterisk boxes and if either one stops the system remains running |
| 07:13.38 | UncleKiwi | ie using carp etc |
| 07:16.51 | ChannelZ | I don't really have any experience with that. |
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| 10:57.28 | pukkita_ | Hi |
| 10:57.42 | pukkita_ | How could it be a simple NoOP can screw a macro??? |
| 11:00.45 | pukkita_ | trying to hack elastix and placing a NoOP() or Verbose() screws the macro with no other debug message that it exited non zero???? |
| 11:07.24 | wdoekes | diff or it didn't happen |
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| 11:08.07 | pukkita_ | wdoekes: what do you mean? |
| 11:08.38 | pukkita_ | it explicitly says the macro exited non zero on that line |
| 11:08.55 | pukkita_ | the line with the NoOP or Verbose, same line in other contexts work fine |
| 11:08.57 | wdoekes | cp extensions.conf{,.orig}; [do changes...]; diff -pu extensions.conf{.orig,} |
| 11:09.20 | pukkita_ | if I comment out the NOOP or Verbose it works fine |
| 11:09.34 | pukkita_ | what puzzles me is how a simple NoOP can screw it? |
| 11:09.48 | wdoekes | stop talking, start showing the diff |
| 11:12.13 | pukkita_ | ok |
| 11:12.17 | pukkita_ | --- extensions_additional.conf2014-11-06 12:11:05.000000000 +0100 +++ extensions_additional.conf.orig2014-11-06 12:10:44.000000000 +0100 @@ -1623,7 +1623,7 @@ exten => s,n,ExecIf($[$["${MOHCLASS}" != exten => s,n(gocall),Macro(dialout-trunk-predial-hook,) exten => s,n,GotoIf($["${PREDIAL_HOOK_RET}" = "BYPASS"]?bypass,1) exten => s,n,GotoIf($["${custom}" = "AMP"]?customtrunk) -exten => s,n,Verbose(** CONNTRUNK: ${CONNECTEDLIN |
| 11:12.21 | pukkita_ | ouch |
| 11:12.23 | pukkita_ | sorry |
| 11:12.31 | Rac-on | put it in a pastebin ;) |
| 11:12.33 | pukkita_ | gonna use pastebin |
| 11:13.42 | pukkita_ | http://pastebin.com/YnYRG9x4 |
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| 11:15.40 | wdoekes | ok, now we believe you. and the Verbose app is loaded? |
| 11:16.21 | pukkita_ | Nov 6 12:00:05] VERBOSE[2889] pbx.c: == Registered application 'Verbose' |
| 11:16.29 | pukkita_ | also it works fine on other contexts |
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| 11:16.34 | wdoekes | try enabling core set debug 5 (or so) and check the debug log |
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| 11:27.23 | pukkita_ | [Nov 6 12:14:38] VERBOSE[16020] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/1006-00000064' in macro 'dialout-trunk' |
| 11:28.07 | pukkita_ | also a reload after commenting out the line doesn't fix it, it needs a restart |
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| 11:52.29 | pukkita_ | ouch |
| 11:52.42 | pukkita_ | tons of debug but cannot find anything related to that macro |
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| 14:03.26 | Ta^3 | newto |
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| 14:20.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 14:35.54 | pa | WIMPy, is the LCR mailing list the one at isdn4linux? |
| 14:36.41 | WIMPy | That's the one to use, yes. |
| 14:37.21 | pa | thanks! :) i will write there, as Jolly never actually answered me, and i decided to eventually go voip and use linux as call router |
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| 14:38.45 | WIMPy | I'm considering to fork it. Staying with the old Asterisk integration. |
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| 14:40.12 | WIMPy | thinks making it incompatible to the standard Kernel was a bad thing. |
| 14:41.47 | WIMPy | Things need to be more user friendly. Or should we call it admin friendly? |
| 14:45.21 | pa | hehe admin i think :) |
| 14:45.55 | pa | i mean, ideally i'd like to have the dahdi-hfcs ported to 3.10+.. to require one less piece of software running |
| 14:46.09 | pa | but since it's abandoned.. |
| 14:46.21 | WIMPy | Is it? |
| 14:46.44 | pa | donno, it looks like. I wrote to the mantainer, but he never answered |
| 14:46.50 | pa | raoul boenisch |
| 14:48.31 | WIMPy | Having an extra application was the reason why I waited so lo to go that way. But for what I do most of the time, it's actually a big advantage. I only route those calls to Asterisk that need to go |
| 14:48.36 | WIMPy | there. |
| 14:49.17 | pa | that's true in a way.. it would be more resistant to asterisk changes.. |
| 14:49.28 | pa | anyway, one way or another would be fine |
| 14:50.02 | WIMPy | What was your issue with lcr? |
| 14:50.54 | pa | i dont get my voice to the other. i get their voice though |
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| 14:51.17 | pa | not sure it's because of missing loopback driver, but it might be |
| 14:51.21 | WIMPy | Using hfcpci? |
| 14:51.26 | pa | yes |
| 14:51.41 | pa | so voice goes LCR -> Asterisk, but not the other way around |
| 14:51.45 | WIMPy | For the current lcr you need the loopback to make it work with Asterisk, yes. |
| 14:52.05 | WIMPy | Err. So you're on the old lcr? |
| 14:52.11 | pa | yours, yes |
| 14:52.16 | pa | the 1.12 |
| 14:52.32 | pa | new one i did not manage to get it to work |
| 14:52.45 | WIMPy | But there was a patch for hfcpci in the mailing list two days ago. That migh be for exactely that issue. |
| 14:52.54 | pa | aha! |
| 14:53.07 | pa | thanks! i'll look at that |
| 14:53.29 | WIMPy | I'm on hfcmulti everywhere, so I haven't tried hfcpci for a long time. |
| 14:53.46 | pa | interesting.. are those cards expensive? |
| 14:54.38 | WIMPy | Well, compared to the single port ones, they are extremely expensive. But compared to other telephony cards, not really. |
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| 14:57.27 | pa | hm i see.. then maybe not. I have only one isdn line after all .. |
| 14:58.25 | pa | the big problem for me is that my phone operator increased the prices tremendously to make everybody move to some bulk deals, but they do not have any deal for old ISDN lines because they do not sell them anymore. And i want to keep it because of the second line. |
| 15:00.52 | pa | so i want to change all the phones to voip phones, to prevent mistakes of calling using the wrong phone, at the crazy fare. :) |
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| 15:01.02 | pa | but for that i need to route ISDN to asterisk :) |
| 15:01.28 | pa | anyway, will check the patch |
| 15:01.33 | WIMPy | Not sure I understand that one. |
| 15:01.53 | WIMPy | Why do you want to exchange phones? |
| 15:03.14 | pa | because if i leave the old pstn phones, and add voip phones for out calls, people might make mistakes and still use the old ones to call.. |
| 15:03.48 | WIMPy | See, that's where the multiport card comes in to play :-) |
| 15:03.49 | pa | anyway, the patch seems for the mISDN driver, but i wonder if its possible to apply it to the version in 3.13.. |
| 15:04.19 | pa | well, i found cheap voip phones, they will do |
| 15:04.24 | pa | like 10⬠or so |
| 15:04.43 | WIMPy | ok |
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| 15:06.11 | ektat | hi there |
| 15:07.29 | carrar | hi |
| 15:08.00 | ektat | After sifting through google results, reading the 4th ed of Asterisk guide and lost a few thousands hairs. I have something to ask :) |
| 15:08.03 | pa | WIMPy, so the new LCR simply does not work? |
| 15:08.20 | carrar | ~ask |
| 15:08.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:08.43 | ektat | is there a way to remove the "party=calling" from Remote-Party-Id |
| 15:09.03 | ektat | without using AddSIPHeader of course |
| 15:09.09 | WIMPy | pa: Not without the loopback module which only exists on git. |
| 15:10.23 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 15:11.19 | [TK]D-Fender | ektat: You can't remove headers |
| 15:12.58 | pa | WIMPy, have you ever tried to use the standalone mISDN driver instead of the kernel one? |
| 15:13.12 | pa | i read it should be possible to have them side by side |
| 15:13.14 | pa | but i tried |
| 15:13.40 | WIMPy | pa: Yes, but that was a long time ago. They often didn;t work with the latest kernel. |
| 15:13.45 | ektat | [TK]D-Fender thanks, there's a way to remove SIP headers actually (SIPRemoveHeader()), but I just want to alter it |
| 15:13.45 | pa | and even with the loop module the new lcr didnt work. but i'm not sure if i should have blacklisted the kernel modules to give priority to the standalone ones |
| 15:14.05 | [TK]D-Fender | ektat: Actrually news to me... I will have to look that up.. |
| 15:14.32 | [TK]D-Fender | ektat: If that app is there then it would seem you'd have to read it, strip it, then readd it |
| 15:14.45 | WIMPy | pa: You can't. They have the same names. So you probably need to remove the original ones. (or not build them) |
| 15:15.06 | ektat | [TK]D-Fender yeah, I thought there was a "cleaner" way but I guess I'll resort to that |
| 15:15.18 | WIMPy | ektat: Do you really want rpid? |
| 15:15.18 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 15:15.31 | [TK]D-Fender | ektat: usually if there is any way to do a thing... don't bet on there being a second ;) |
| 15:15.45 | ektat | WIMPy Sure my trunk provider wants this |
| 15:15.49 | [TK]D-Fender | ektat: I could name a few exceptions though... |
| 15:16.05 | ektat | [TK]D-Fender asterisk docs are so poor I had to ask anyway |
| 15:16.31 | WIMPy | I have replaced all rpid by pai a long time ago. |
| 15:17.01 | ektat | WIMPy I'd love to but these suckers from the provider don't use pai |
| 15:17.07 | [TK]D-Fender | ektat: Well you did see that app there.. not sure what exactly qualifies as "poor" |
| 15:17.36 | ektat | [TK]D-Fender I meant it's very difficult to find examples and recipes |
| 15:18.02 | ektat | especially if you're not on 1.4-1.8 |
| 15:18.11 | carrar | ektat: You could do it outside of asterisk |
| 15:18.14 | carrar | opensips |
| 15:18.23 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 15:18.33 | [TK]D-Fender | ektat: Ah... well * is largely the same as regular programming really. We can show the syntax, but it's up to the user to deploy it in a way that''s meaningful to them. |
| 15:18.54 | [TK]D-Fender | ektat: I use * as a jukebox and coffee-maker.... |
| 15:18.59 | [TK]D-Fender | ektat: YMMV |
| 15:19.08 | carrar | And Oh what YUMMIE coffee it makes |
| 15:19.22 | [TK]D-Fender | ektat: Oh, and there is a book or two on "recipes" for * |
| 15:19.40 | carrar | With a coffee-maker, sound card and a soldering iron you can make a ATA |
| 15:19.51 | [TK]D-Fender | ~savemoney |
| 15:19.51 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 15:19.56 | ektat | [TK]D-Fender jukebox is done here too :) |
| 15:19.57 | [TK]D-Fender | Never gets old.. just older ;) |
| 15:20.00 | carrar | never |
| 15:20.01 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 15:20.03 | pa | WIMPy, it is not enough to blacklist them? |
| 15:20.10 | pa | (the kernel modules i mean) |
| 15:20.31 | pa | oh same name you say.. |
| 15:20.32 | pa | indeed |
| 15:20.49 | pa | maybe then i need to rebuild the kernel without mISDN, and try only the standalone one |
| 15:21.03 | pa | (which now would even have those patches) |
| 15:21.09 | ektat | btw, thanks for all your suggestions |
| 15:22.05 | [TK]D-Fender | ektat: http://it-ebooks.info/book/619/ |
| 15:22.50 | ektat | hey thanks already read this one, not very useful to me |
| 15:23.02 | [TK]D-Fender | ektat: http://www.chapters.indigo.ca/books/product/9780321525666-item.html?s_campaign=goo-PLATest&gclid=CNKC-qGk5sECFVgOjgodAEIA8g |
| 15:23.17 | [TK]D-Fender | ektat: both kinda old, but may have newer versions out. |
| 15:23.41 | [TK]D-Fender | ektat: You can alsways ask in here if you hit a bump on things you are trying to implement |
| 15:24.05 | WIMPy | pa: That's probably the way if you want to upgrade lcr. |
| 15:24.34 | carrar | ektat: have you tried to save the header, then remove it, edit the saved captured header, then add it? |
| 15:25.15 | ektat | carrar that's the only solution I came up, I thought there was some kind of parameter switch to do this in a clean fashion |
| 15:26.02 | WIMPy | ektat: I'd disable the rpid headers and add them manually. |
| 15:26.19 | carrar | Asterisk have never been big on SIP header manipulations commands |
| 15:26.49 | ektat | carrar I didn't try the new pj_sip driver, is it better ? |
| 15:26.51 | *** join/#asterisk wizhippo (~wizhippo@64.201.57.7) |
| 15:26.55 | carrar | thats where having something like opensips/kamilio comes in handy |
| 15:27.25 | carrar | ektat, I haven't done much with pj_sip yet |
| 15:29.01 | ektat | I'll do it the "dirty" way (store,remove,rewrite,add)...nevermind |
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| 15:30.22 | pa | WIMPy, well i just want it to work :) if that makes it work, i'll do it, because as far as i see, it's probably the easiest way to get the patch too. I don't think it will be easy to backport that patch to 3.13 version |
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| 15:30.37 | pa | but i will try next week when i'm physically at the server :) |
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| 15:33.36 | WIMPy | pa: I don't see why you shouldn't be able to apply that to whatever kernel you use. |
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| 15:39.42 | ektat | sorry for that autoaway crap |
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| 15:52.06 | jamesc | Anyone know what controls asterisk retransmits ? |
| 15:52.55 | WIMPy | What controls? From where? to where? |
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| 15:53.25 | jamesc | When I make a call the invite get retransmitted 3 times really quickly |
| 15:53.26 | WIMPy | needs something to eat. |
| 15:54.00 | WIMPy | Then it didn't get an answer. |
| 15:54.26 | jamesc | the call progresses as normal though |
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| 15:55.01 | jamesc | I am seeing retransmits in the impossible to reply that fast time frame |
| 15:55.43 | WIMPy | Did you screw some timer configuration? |
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| 15:57.33 | jamesc | Thats the sort of thing but I have not any wired timer config |
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| 16:19.20 | jamesc | Asterisk seems to send 2 retransmits for an initial invite consecutively. after 0.0004 seconds this is too quick I think. Is there a config parameter that controls this? |
| 16:20.31 | [TK]D-Fender | Show us. |
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| 17:20.40 | jamesc | Ok what is the best way to show you i have a small pcap file? |
| 17:20.48 | jamesc | [TK]D-Fender: ^ |
| 17:21.19 | [TK]D-Fender | * CLI w/ SIP debug |
| 17:21.29 | [TK]D-Fender | ~pb |
| 17:21.29 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:21.31 | [TK]D-Fender | ^ |
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| 17:40.16 | Qwell | zomg tuxd00d |
| 17:40.44 | tuxd00d | Hey Qwell ! :) |
| 17:41.01 | Qwell | tuxd00d: Driven any strangers to SF lately? |
| 17:41.02 | tuxd00d | The podical son has returned ;) |
| 17:41.49 | tuxd00d | Oh, not lately. And weâre part of the same asterisk brotherhood, so not a stranger. :) |
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| 18:34.48 | casdude | hey, I have a call recorder that receives a call in from the telco and then forward the call on to a 3rd party PBX, this seems to work fine however the leading zero is always missing. |
| 18:34.55 | casdude | My extensions.conf http://pastebin.com/FyQR3cN6 and extensions_additional.conf http://pastebin.com/yb1MHc8Q look like this and my extensions_customer.conf http://pastebin.com/rb8k9MN5 looks like this. Just wondering if anyone knew where I was going wrong or why my change is not taking effect, although simple it looks correct to me. |
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| 18:36.03 | WIMPy | What leading 0? From where? And where do you miss it? |
| 18:38.51 | casdude | hi Wimpy this is a line from the console to hopefully show, the number dialed is 01633... however the zero has been removed 2014-11-06 12:18:39] VERBOSE[28798][C-00000000] pbx.c: -- Executing [tdial@ext-trunk:5] Set("DAHDI/i1/1633508781-1", "DIAL_NUMBER=642311") in new stack |
| 18:39.20 | casdude | when this call is then placed to the PBX the number is 1633.... |
| 18:39.34 | casdude | which is wrong it should be 01633 |
| 18:39.53 | WIMPy | Where has it been dialed? On a PBX connected to your Asterisk? |
| 18:40.24 | casdude | from external through the telco into the asterisk system then on to the pbx |
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| 18:41.05 | WIMPy | So that is the called number of a call from the telco? |
| 18:41.17 | casdude | yes |
| 18:41.31 | [TK]D-Fender | casdude: The custom bit of dialplan you showed changes the CALLERID, not the CALLED NUMBER. |
| 18:42.06 | casdude | ok |
| 18:43.17 | WIMPy | A nmumber starting with a 0? |
| 18:43.45 | casdude | Wimpy: yes a number starting with |
| 18:43.49 | casdude | a zero |
| 18:43.49 | WIMPy | Or is that including an area code which would be prefixed by a 0? |
| 18:44.51 | casdude | the inbound call is made from 01633... this is then forwarded to the PBX however the number is appearing as 1633 |
| 18:45.10 | WIMPy | Now you say from. |
| 18:45.34 | WIMPy | Is it the calling or the called number? |
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| 18:45.58 | WIMPy | >>Or is that including an area code which would be prefixed by a 0? |
| 18:46.00 | casdude | the calling number |
| 18:46.27 | WIMPy | These prefixes are not part of the number. |
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| 18:49.19 | casdude | they are there when the system is not in place, the pbx reports the number fully |
| 18:49.34 | casdude | this is a line from the PBX |
| 18:49.49 | WIMPy | It adds the zero(s) itself. |
| 18:50.01 | casdude | when the recorder is not in place |
| 18:50.06 | casdude | <PROTECTED> |
| 18:50.20 | WIMPy | You should be able to configure chan_dahdi to add them, but make sure to remove the flas then, otherwise the PBX will double them. |
| 18:50.24 | casdude | <PROTECTED> |
| 18:50.30 | casdude | when the system is in place |
| 18:50.49 | WIMPy | How many boxes are we talking baout? |
| 18:50.54 | WIMPy | about |
| 18:51.24 | casdude | [telco] > [recording server asterisk] > [PBX asterisk] |
| 18:51.59 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-110-145.dynamic.qsc.de) |
| 18:53.53 | WIMPy | Hmm. That doesn't add up for me. There is a difference on your Asterisk depending on with or without recording server, but that recording server is the Asterisk we're talking about? |
| 18:54.05 | WIMPy | Or is that another box in front of it? |
| 18:54.57 | casdude | no there is no other box in front of it, it is [telco] > [recording server asterisk] > [PBX asterisk] |
| 18:55.29 | WIMPy | Oh, so it's two Asterisks. And the change is on the 2nd Asterisk. |
| 18:56.54 | casdude | the 2nd box needs to show a leading zero |
| 18:57.01 | WIMPy | And all connections are PRIs? |
| 18:57.04 | casdude | yeh |
| 18:57.16 | casdude | i am happy to make it on which ever box tbh :) |
| 18:58.17 | WIMPy | Ok, so the 1st one does seem to do some search/replace then. Did you configure any prefixes/are codes or so on the 1st one? |
| 18:58.34 | WIMPy | I guess that might remove some information. |
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| 18:58.54 | casdude | agreed |
| 18:59.09 | casdude | the strange thing is I cant see where or why |
| 18:59.36 | casdude | I have been trying to get it to add a zero back in but no luck so far |
| 18:59.39 | WIMPy | And the *dialplan parameters to dynamic? |
| 18:59.40 | [TK]D-Fender | Perhaps you should show us then |
| 19:00.29 | casdude | within chan dahdi? |
| 19:00.41 | WIMPy | yes |
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| 19:02.14 | casdude | its not set, i can add. |
| 19:03.41 | WIMPy | Whatever that would make by default. |
| 19:04.47 | casdude | ok i have set to host=dynamic |
| 19:04.58 | casdude | unfortunately no change |
| 19:05.13 | WIMPy | host? What host? |
| 19:05.20 | WIMPy | pri*dialplan |
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| 19:06.16 | casdude | sorry i thought setting to dynamic would be host=dynamic to chan_dahdi |
| 19:07.02 | casdude | what where you thinking of needing to be set to dynamic |
| 19:07.27 | WIMPy | pri*dialplan |
| 19:12.19 | casdude | ok it is set to dynamic |
| 19:12.28 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:12.42 | casdude | unfortunately no change |
| 19:13.56 | WIMPy | Then show us a call with debug for both spans enabled and the chan_dahdi.conf. |
| 19:14.20 | WIMPy | And verbose >=3, as always. |
| 19:15.24 | marceloamorim | hello guys, at Asterisk 11.9.0 in chan_dahdi.conf should I use every parameters bellow the context=x or could I use commons settings above the first context and those settings was set for every channel, or should I repeat the settings for every channel? I have some issues that seems related |
| 19:16.22 | WIMPy | All settings apply to all following channels. |
| 19:17.05 | WIMPy | bbiab |
| 19:19.38 | marceloamorim | ok then, ty WIMPy |
| 19:20.15 | casdude | hey here is the verbose and debug out put http://pastebin.com/VgXwmdQN |
| 19:20.28 | casdude | and the chan_dahdi settings |
| 19:20.36 | casdude | http://pastebin.com/xctiaEBw |
| 19:21.17 | casdude | thanks for taking the time to look, it is really appreciated |
| 19:25.15 | WIMPy | Ok, so that's a ratehr short chan_dahdi.conf. It has no channels. So that's obviousely not complete. |
| 19:25.37 | WIMPy | But I do see a nationalprefix=0 there. So that might already be the one that eats the 0. |
| 19:26.26 | [TK]D-Fender | PRI Span: 1 < Presentation: Presentation allowed, User-provided, not screened (0) '1633508781' ] |
| 19:26.43 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 19:26.48 | casdude | i just added that, the channels are stored in /dahdi/system.conf on this setup |
| 19:27.01 | WIMPy | You can see the TON change from national to unknown. |
| 19:27.01 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 19:27.34 | WIMPy | And what does that file contain? |
| 19:30.10 | casdude | pass i dont know what the unknown file contains |
| 19:30.17 | casdude | i did not configure it |
| 19:30.28 | casdude | to do anything |
| 19:30.59 | casdude | i just rang the same test on the pbx with out the call recorder in place and received the following http://pastebin.com/hYD4F9Y2 |
| 19:31.09 | casdude | not sure if it helps out at all |
| 19:31.13 | WIMPy | The channels must be configured somewhere. Otherwise you wouldn't be able to get a call. |
| 19:31.56 | WIMPy | No, it's the same that was received in the other PB. |
| 19:32.05 | WIMPy | But what you send out is obviousely different. |
| 19:32.32 | casdude | yeh by the location where the channels are configured do you mean this http://pastebin.com/6TmBmx9x |
| 19:32.37 | WIMPy | And you should try to set all pri*dialplan parameters to dynamic. |
| 19:32.45 | casdude | located in /etc/dahdi/system.conf" |
| 19:32.56 | WIMPy | Nope. |
| 19:33.10 | WIMPy | Something in asterisk/chan_dahdi.conf or included there. |
| 19:34.14 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:34.53 | casdude | arhhh i have it http://pastebin.com/W48uTqM1 |
| 19:35.12 | casdude | this is located in /etc/asterisk/chan_dahdi_groups.conf on this build |
| 19:35.58 | WIMPy | Ok. That's definitely going to cause trouble. |
| 19:36.44 | WIMPy | There the pri*dialplan's are set to unknown. That *will* clear the flag you're missing. |
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| 19:42.49 | casdude | i have set to dynamic and am still having an issue |
| 19:42.56 | casdude | having the issue |
| 19:43.16 | casdude | i tried, local and international while i was testing also |
| 19:43.39 | WIMPy | Don't set it to a fixed value. |
| 19:43.53 | WIMPy | That's almost crtainly wrong. |
| 19:45.02 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 19:45.58 | WIMPy | Did you remove all *prefix parameters and set all pri*dialplan parameters to dynamic? |
| 19:47.59 | casdude | tes |
| 19:48.02 | casdude | yes |
| 19:49.10 | WIMPy | And you reloaded chan_dahdi? Or maybe better restarted Asterisk? |
| 19:53.05 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
| 20:00.06 | *** join/#asterisk _0x1d3 (~0x1d3@2.216.128.25) |
| 20:02.08 | WIMPy | Oh, and as far as FreePBX generating that config goes, I'd consider that a bug. |
| 20:03.57 | casdude | sure |
| 20:05.27 | casdude | it appears to be working, kinda, now there are two leading zeros, thanks for all your efforts. unfortunately the cleaner is about to leave and my dinner is probably in the dog by now |
| 20:06.01 | casdude | you have definately pointed me in the right direction thanks for all your help again |
| 20:06.28 | casdude | bye for now :) |
| 20:07.29 | WIMPy | Did you set something to national on span 2? |
| 20:09.49 | *** join/#asterisk areski (~areski@cust114-dsl56.idnet.net) |
| 20:15.24 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
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| 21:18.55 | Micc | it seems like asterisk is ringing all available lines of a phone in the queue. Is the only way to prevent this to set call-limit in sip.conf? |
| 21:19.06 | Micc | This is in 1.8.9.2 |
| 21:25.32 | Micc | It seems that call-limit will limit the number of outbound calls as well. I just want to ring once per available device in the queue but allow that phone to use other lines for dialout. |
| 21:26.30 | Micc | now its still trying to call but its rejecting due to usage limit of 1. |
| 21:26.49 | Micc | But I guess I don't understand why its ringing the same device 6 times in the first place. |
| 21:27.05 | Micc | maybe this is a bug that has been fixed in newer versions? |
| 21:32.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jctqrhidjugpdyhn) |
| 21:32.38 | Penguin | You've either configured the different lines as members of the queue or you've configured all the line keys with the same SIP account. |
| 21:32.51 | Penguin | Both would be wrong. |
| 21:34.18 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 21:39.06 | Micc | The device is only registering once, but can use up to 6 lines. |
| 21:39.28 | Micc | I didn't have this problem when I was using static members. |
| 21:39.40 | Penguin | What are you using now? |
| 21:40.04 | Micc | I'm using AddQueueMember |
| 21:40.18 | Micc | each phone logs in when they want to join the queue. |
| 21:42.57 | *** join/#asterisk jhlavacek (~jirka@jix.nextadsl.cz) |
| 21:44.04 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 22:00.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:03.17 | *** join/#asterisk nny (~Scott@cpe-174-107-201-051.sc.res.rr.com) |
| 22:04.09 | nny | Odd issue... I thin it may be a setting in the polycomm config. I am still troubleshooting. Whenever I use externhost vs externip my polycom remote phone becomes UNREACHABLE. This doesn't happen on the local network. The host name is resolvable on the remote network. |
| 22:05.23 | Penguin | But is the host name resolvable where it counts: on the network where asterisk resides? |
| 22:07.06 | Penguin | Asterisk has to look up the host name that you set in externhost, to apply it to an implied externaddr. If asterisk cannot correctly resolve the externhost value, the rest is a failure as well. |
| 22:07.59 | Penguin | 'sip show settings' should show you the host name and resolved address. |
| 22:08.09 | *** join/#asterisk areski (~areski@cust114-dsl56.idnet.net) |
| 22:19.15 | *** join/#asterisk blackslikz (~blackslik@81.17.25.66) |
| 22:19.53 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 22:21.28 | nny | Penguin: ahh. The dhcp updated /etc/hosts to the local network. I don't use that DNS for the asterisk normally as it has the externhost set to a local IP for non remote phones. You win! |
| 22:21.41 | nny | So it was resolving to the internal IP :\ |
| 22:23.34 | nny | PEERDNS=no :\ They use dhcp to assign the static ip. Fixed. Thanks Penguin |
| 22:23.51 | nny | someone needs to make a way to tip people with bitcoin etc here :) |
| 22:25.40 | blackslikz | hello All .. i have a major issue with my asterisk .. its been runing for months .. right now if i run asterisk it says [root@host ~]# asterisk -rvv |
| 22:25.40 | blackslikz | -bash: asterisk: command not found |
| 22:28.55 | [TK]D-Fender | Check your path, then start hunting for the files |
| 22:34.57 | *** join/#asterisk asterisk562 (d4b2f61f@gateway/web/freenode/ip.212.178.246.31) |
| 22:35.17 | asterisk562 | anybody? |
| 22:36.33 | asterisk562 | knock, knock, any human here? |
| 22:36.34 | *** join/#asterisk bkruse (~Adium@173-14-196-25-huntsville.hfc.comcastbusiness.net) |
| 22:37.13 | newtonr | ~ask |
| 22:37.13 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:38.05 | asterisk562 | banging my head last few days, how to detect channel volume increase, say, 30% |
| 22:38.46 | asterisk562 | tried rtcp debug from cli, no luck |
| 22:39.44 | asterisk562 | any idea? |
| 22:41.02 | newtonr | asterisk562, no idea! |
| 22:41.40 | newtonr | If no one has an idea here, you might post on the asterisk-users list. |
| 22:42.05 | asterisk562 | thanks anyway, any other channel here I should visit to get more info? |
| 22:44.03 | [TK]D-Fender | asterisk562: You'd have to make some sizable mods to *. |
| 22:44.22 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
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| 22:45.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:45.49 | asterisk562 | I can change C code and recompile *, just asking here before start digging |
| 22:46.40 | asterisk562 | perhaps I should try with some rtp analyzer? |
| 22:48.15 | [TK]D-Fender | asterisk562: Any outside RTP analysis... is an outside solution |
| 22:48.34 | [TK]D-Fender | asterisk562: As for * ... you've got the source, you can do with it whatever you will (with GPL in mind) |
| 22:49.09 | newtonr | asterisk562, #asterisk-dev if you need to ask questions about the source |
| 22:49.23 | asterisk562 | thought I can use rtp analyzer to trigger AMI action |
| 22:49.57 | asterisk562 | any recommendation for rtp analyzer? |
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