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| 09:33.08 | minghags | yes this i know, but can i maybe disable "tcptls" module somewhere in "menu select"? because I dont need it at all... |
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| 17:37.29 | znoteer_ | incoming calls to my newly-ported-to-voip number are not making it through to my pbx. Setting verbosity to 10 and iax debugging on, I can see a REJECT message with cause noted as "No authority found". I don't know where to go from here. Searching the web has turned up some hits, but none that match my situation close enough to be useful. Can anyone help me to solve this? |
| 17:39.12 | znoteer_ | I actually saw one hit that was somewhat similar. The problem was authentication credentials that were wrong. I've checked my username and password on my end and the provider's end, and they match, so that's not the problem |
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| 21:42.02 | znoteer_ | I've just switched from PSTN to voip for incoming calls. When my number is called it is automatically answered and the user is asked to choose a language. While my number was with the PSTN there was no problem with the playback of this prompt. Now that I'm using voip, the first word of the prompt is cut off. Is there some way to make asterisk wait a second between the Answer() and the Playback() of my prompt? I've looked at wait and pause, and tried to |
| 21:46.37 | kitsune | Playback(silence/1&my_prompt) |
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| 22:04.47 | znoteer_ | kitsune: Thanks! that did the trick. Any idea where this kind of thing is documented? There's nothing in the Oreilly book nor in the command reference of the asterisk.org wiki |
| 22:05.51 | znoteer_ | Oreilly book 3rd edition at any rate |
| 22:08.09 | kitsune | It's in the 4th edition for sure. Also under Playback() in the wiki. |
| 22:12.12 | kitsune | It's one of those Asterisk "idioms", it should usually be done by default. |
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| 22:27.41 | [TK]D-Fender | That isn't a "thing", that's just playing a sound file that's included. |
| 22:27.51 | [TK]D-Fender | You should look at what's in your sounds folder |
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| 22:49.51 | sibiria | kitsune: Answer(ms) is perhaps better |
| 22:50.15 | sibiria | will answer the call and wait that many milliseconds before proceeding |
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| 22:51.14 | [TK]D-Fender | Generally I'd disagree. the extra moment is often benficial in making sure RTP is actually fully set up so that audio doesn't get truncated at the start fo r the announcement |
| 22:51.56 | sibiria | that problem is common, and known to anyone with a mobile phone. we actually made a change in our DID portion just this week to handle that problem |
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| 22:52.29 | sibiria | and we came to the conclusion that just delaying before playing audio solved it just the same as trying to "kickstart" the RTP channel by first playing some silence to get data out |
| 22:53.20 | sibiria | thus, Answer(delay) worked just as good as Answer + Playback(en/silence/1) |
| 22:53.35 | sibiria | mileage may vary, of course |
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| 23:01.13 | kitsune | That makes sense though. |
| 23:02.02 | kitsune | Are you behind a NAT? |
| 23:02.42 | sibiria | part of the infrastructure is |
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| 23:23.31 | kitsune | Are there any good tools or interfaces for monitoring call quality or internal mechanisms related to call quality like jitter buffers on asterisk? |
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| 23:36.30 | TurBoss | Hello |
| 23:36.31 | TurBoss | I have configured my sip setting in pjsip_custom.con |
| 23:36.31 | TurBoss | now i see it registered |
| 23:36.32 | TurBoss | how I can route incoming calls to a ringer gruop¿ |
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| 23:47.55 | TurBoss | this -> PJSIP/$OUTNUM$@mycustomtrunk ? |
| 23:52.47 | [TK]D-Fender | That isn't a "ring group" |
| 23:52.59 | TurBoss | ah yes |
| 23:53.21 | TurBoss | I needed to register a trunk first |
| 23:53.23 | TurBoss | :) |
| 23:53.24 | TurBoss | sorry |
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