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| 01:34.34 | wakko | hi, first JIRA submission: https://issues.asterisk.org/jira/browse/ASTERISK-25041 comments welcome |
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| 02:19.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:53.09 | [sr] | ai |
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| 10:20.40 | [sr] | working day |
| 10:20.47 | [sr] | everyone should be working!! |
| 10:20.48 | [sr] | :p |
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| 11:43.21 | Zogot | hey there, where is asterisk-sounds rpm these days? |
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| 11:57.09 | ipalmer | Hi all, I'm having trouble receiving calls from a Cisco pbx to my Asterisk box via SIP, I'm using chan_sip not pjsip. There is no audio when I answer the call, it's all internal so shouldn't be NAT related. I get an error from the asterisk cli saying 'Failing due to no acceptable offer found' google says it is because the m=port number is 0 but my sip logs don't show this, could someone look at the sip logs and see if they |
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| 13:12.05 | skyroveRR | So I tried chan_mobile which seems to be officially supported by the asterisk community on asterisk version 1.8.24.0. When I try to make the dialplan recognise DTMF tones, asterisk seems to detect them only once for every 20 calls on average. Any fixes for this? |
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| 13:21.34 | eschmidbauer | I've heard much of the core of Asterisk was rewritten between version 1.8 and 11? Is this true? |
| 13:21.43 | eschmidbauer | Is there a changelog for these core changes? |
| 13:22.02 | [TK]D-Fender | It's in this file named ....."changelog" |
| 13:22.11 | eirirs | <PROTECTED> |
| 13:22.31 | [TK]D-Fender | supplented by this other file named ..... "changes" |
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| 13:22.56 | eschmidbauer | dick |
| 13:23.47 | eschmidbauer | changelog doesn't show what are core changes and what are not |
| 13:23.57 | eschmidbauer | i dont really care about a change to sounds.xml file |
| 13:25.05 | [TK]D-Fender | both files cover what is distributed as documentation |
| 13:25.09 | [TK]D-Fender | That is what there is |
| 13:25.20 | [TK]D-Fender | Short of reading every JIRA patch |
| 13:25.30 | [TK]D-Fender | And be careful with the name-calling |
| 13:26.02 | skyroveRR | So I tried chan_mobile which seems to be officially supported by the asterisk community on asterisk version 1.8.24.0. When I try to make the dialplan recognise DTMF tones, asterisk seems to detect them only once for every 20 calls on average. Any fixes for this? |
| 13:26.03 | [TK]D-Fender | You also skipped over an entire version |
| 13:26.04 | eschmidbauer | I was hoping someone could verify what i've heard, that the core was rewritten and perhaps provide some details |
| 13:26.11 | eschmidbauer | right 1.8 -> 10 |
| 13:26.20 | WIMPy | There was the opaquification thing going on. But I'm not sure that counts as a major rewrite. |
| 13:26.24 | eschmidbauer | the changelog is not helpful |
| 13:26.25 | [TK]D-Fender | Not "the entire core" |
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| 14:04.08 | ipalmer | Hi all I have an error saying 'chan_sip.c failed due to no acceptable offer found' googling indicated this was an error due to codec mismatch but I am allowing all codecs, anyone able to help with this |
| 14:06.36 | [TK]D-Fender | ipalmer: go look at the actual offer |
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| 14:09.16 | ipalmer | D-Fender: thanks, I have been looking through the SIP debug and the sdp but can;t see anything untoward, m=audio 0 was what I was supposed to look for but all my m=audio messages had a port number |
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| 17:38.49 | redrook | anybody here with info about asterisk 13+ webrtc + video ? |
| 17:39.48 | [TK]D-Fender | Just ask your specific question about it |
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| 18:19.12 | xtuh | hi, is it a good idea to use asterisk for 20 users inside esxi vm |
| 18:20.49 | [TK]D-Fender | IT can work |
| 18:21.09 | [TK]D-Fender | That should be a very tiny load |
| 18:23.26 | xtuh | i need to be shure, what to buy... 2cpu xeon 3,3 ghz for esxi or something just enough but without virtualization, |
| 18:23.53 | [TK]D-Fender | Overkill |
| 18:24.02 | [TK]D-Fender | My Watch can handle this... and it's ANALOG |
| 18:24.37 | WIMPy | wants to see how you istall * on that. |
| 18:25.30 | xtuh | thanks, is there something i need to attention before i get started? |
| 18:25.51 | [TK]D-Fender | ~book |
| 18:25.51 | 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 |
| 18:25.53 | [TK]D-Fender | ^^^ |
| 18:26.00 | [TK]D-Fender | Go get started |
| 18:27.30 | wanda_ | Hi everybody :) I let a message on asterisk.org forum about a BarCamp I coordonated some days ago |
| 18:28.41 | wanda_ | I shared the doc on this website http://barcamp.org/w/page/95337185/BarCampIRC-VoIP-opensource and the .pdf (meeting summary) his here https://twitter.com/v_dagrain/status/594130487247003648 |
| 18:28.52 | wanda_ | I wanted to wish you a goooood week-end :) |
| 18:29.10 | xtuh | what distro is better? freebsd, debian, centos, ubuntu? |
| 18:29.25 | [TK]D-Fender | xtuh: Any Linux you can properly administer |
| 18:30.14 | Chainsaw | Fender is right. Pick one you know. |
| 18:30.15 | xtuh | freebsd vs linux - linux is preferable? |
| 18:30.33 | Chainsaw | Or pick one that most of your colleagues or friends use, so you can most easily get help. |
| 18:30.38 | [TK]D-Fender | xtuh: Yes |
| 18:30.45 | xtuh | or doesnt matter? |
| 18:30.49 | WIMPy | Use whatever you're comfortable with. |
| 18:31.06 | [TK]D-Fender | * was built for Linux, esp with things like DAHDI involved. |
| 18:31.13 | [TK]D-Fender | Just pick one and go |
| 18:32.20 | xtuh | thanks. what about webgui - is itas separate package or builtin? |
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| 18:32.47 | WIMPy | 3rd party stuff |
| 18:33.31 | xtuh | what is better to use webgui or not? |
| 18:33.39 | [TK]D-Fender | <PROTECTED> |
| 18:34.50 | xtuh | thanks, but why not? limitations or bugs? |
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| 18:34.56 | [TK]D-Fender | Limitations |
| 18:35.01 | [TK]D-Fender | And bugs |
| 18:35.10 | xtuh | :) |
| 18:35.18 | WIMPy | Gives you brain cancer |
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| 18:36.19 | [TK]D-Fender | Worrying about cancer causes ulcers |
| 18:37.25 | Tachoh | ulcers can cause cancer |
| 18:38.17 | xtuh | how traffic will go in that case: server with asterisk located in the main office. there is remote office. internal call is entered on multiline, ip already, then operator transfer call to remote office |
| 18:39.18 | Tachoh | i am pretty sure i have brain cancer from my days of sitting in front of a CRT all day |
| 18:39.28 | [TK]D-Fender | Your description is very hard to read and you haven't given any details about what is talking and how. |
| 18:39.37 | [TK]D-Fender | Tachoh: Still using CRT? Geez... |
| 18:39.39 | xtuh | if remote user cannot answer (by timeout) call must be redirected to celular. |
| 18:40.01 | [TK]D-Fender | xtuh: So go do it |
| 18:40.49 | WIMPy | is still using CRT |
| 18:40.53 | xtuh | ill try to rephrase. |
| 18:41.50 | WIMPy | But I have hopes that decent LCDs will become available some time. The ones in laptops are fine, just the desktop versions suck. |
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| 18:44.59 | xtuh | we have multichannel line that can handle 6 calls at the same time. then we want redirect call to some internal user and if he cant answer, then that incoming call must be redirecter to his cellphone. that possible, right? |
| 18:47.12 | xtuh | and when he will talking traffic will go to asterisk and out of asterisk without any activity to internal ip phone of that user? |
| 18:47.20 | [TK]D-Fender | You can BRIDGE a call out just fine. |
| 18:47.43 | [TK]D-Fender | As for directly redirecting and FREEING the inbound channel... that depends on the technology you are using and your provider |
| 18:47.55 | [TK]D-Fender | You have NOT told us what kind of line this is |
| 18:48.28 | xtuh | its vlan |
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| 18:49.15 | xtuh | so incoming call already ip. not analog |
| 18:49.39 | [TK]D-Fender | IP is VERY vague |
| 18:49.46 | [TK]D-Fender | what EXACT PROTOCOL? |
| 18:51.49 | xtuh | i dont know yet. contract will start in 2 weeks. at this time i just know that ill get vlan with 6 incoming lines with one number. |
| 18:52.57 | Tachoh | xtuh: good luck |
| 18:53.05 | [TK]D-Fender | xtuh: Well we can't answer that until we know for sure. |
| 18:53.13 | [TK]D-Fender | At a minimu you can BRIDGE the call out. |
| 18:53.18 | [TK]D-Fender | But this will use up 2 channels |
| 18:53.42 | xtuh | incoming call -> remote segment of network -> redirect to cellphone... |
| 18:54.52 | [TK]D-Fender | I've answered this twice now... |
| 18:55.10 | xtuh | after redirect, remote location will not get any traffic with that call, right? and 2 of incoming lines will be used. am i understand right? |
| 18:55.52 | [TK]D-Fender | The 2nd site depends on how you set it up. |
| 18:56.01 | [TK]D-Fender | maybe |
| 18:56.35 | xtuh | thanks |
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| 21:15.21 | voip | wonderworld |
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| 22:10.46 | UncleKiwi | hey people how do i change the sounds on asterisk to be en_GB |
| 22:10.55 | UncleKiwi | is that done in the asterisk.conf |
| 22:10.57 | UncleKiwi | ? |
| 22:37.32 | UncleKiwi | opps got it sorted |
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