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00:39.01 | F2Knight | Per [TK]D-Fender instructions http://pastebin.com/UxihB9BH unload reload of module and dial plan dumps, relating to Queues not ringing agents. (Invalid) on status. |
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01:00.09 | F2Knight | [TK]D-Fender what do you think? http://pastebin.com/HNZ0DpPr I removed everyone from the queue. Then added them at the CLI and it loads correctly. Still think this isn't an issue with FreePBX? |
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10:12.29 | stefan27 | how can I make asterisk attach b:AS=512 to the video mediatype part of the SDP offer generated by Dial(SIP/A) |
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11:04.03 | mohan__ | i want to alias outgoing number instead the actual one, i think this can be done with setcallerid but not too sure can you please suggest |
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18:07.42 | lvlinux | Asterisk can proxy g729 wihtout a license as long as it doesn't ever have to do any encode/decode right? |
18:11.38 | [TK]D-Fender | "passthrough" |
18:11.43 | [TK]D-Fender | correct |
18:11.55 | lvlinux | What happens if there is no g729 license and during a call Asterisk decides to encode or decode something (record, play a prompt, etc.)?? Does the call get dropped, the operation just fails and the call continues, or what? |
18:12.25 | [TK]D-Fender | a few things may just not make sound, others = hard drop |
18:12.28 | [TK]D-Fender | most = drop |
18:13.17 | lvlinux | k thanks. I think we'll just continue to avoid g729 then :-) |
18:14.44 | lvlinux | I have a client that I am converting from TDM to VoIP and he was concerned about bandwidth usage on their 10Mb connection. I don't think they have that many concurrent calls though to warrant compression. |
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18:53.04 | txt23 | Hey guys. Fairly new to VOIP. In my SIP phone I have "Music on Hold URI". Does this mean I can put in a link to an MP3 url in there? |
18:54.17 | [TK]D-Fender | It's a SIP requuest for a call processing server. |
18:54.23 | [TK]D-Fender | Which you can forget with * |
19:01.18 | txt23 | So I used sip:music@iptel.org and it worked. :) |
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19:03.24 | [TK]D-Fender | If they host openly like that and you want them trying to reinvite outt, then you can try that... |
19:03.41 | [TK]D-Fender | but it leaves your network and I'm not sure how it'll react if it fails |
19:03.55 | [TK]D-Fender | And keeping in mind that only effects 1 side |
19:04.30 | [TK]D-Fender | (the one configured to point to it themselves. ie: you put me on hold, and I heard what I want... not what they want me to hear |
19:05.00 | txt23 | Yeah I think hosting it internally would be good. Let me try and see if I can learn that |
19:05.49 | [TK]D-Fender | ~book |
19:05.49 | 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 |
19:05.54 | [TK]D-Fender | And read the sample config for it |
19:07.33 | Tachoh | we have 2 asterisk servers writing CDR to the same table in a sql server database using the freetds module. in the logs it says we are experiencing deadlocks, which causes the write to fail and retry. is setting up two asterisk servers to write CDR to the same database a bad idea? |
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23:02.20 | onculayin | How can I improve sound quality of Android clients? I can make and take calls with the phone adapter very well however my android extension clients have terrible sound quality. I tried Linphone annd CSipSimple |
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23:12.54 | [TK]D-Fender | Software generally isn't going to make things better |
23:13.13 | [TK]D-Fender | bad audio is either a bad mic, or bad network |
23:13.38 | [TK]D-Fender | jitter + PL = bad |
23:17.25 | onculayin | [TK]D-Fender: thanks for the info. Would buying the codec from Asterisk help? |
23:18.08 | [TK]D-Fender | Depends if you're choking on BW. |
23:18.22 | [TK]D-Fender | But if that's your issue... the tiny savings is unlikely to save you anything |
23:18.27 | [TK]D-Fender | again... NOT software |
23:18.52 | [TK]D-Fender | unless one does slightly better PLC than others. |
23:18.59 | onculayin | [TK]D-Fender: I see. So basically as long as I am able to make connection means things are working fine on the Asterisk side |
23:19.00 | [TK]D-Fender | but you're always at the mercy of your networking |
23:19.28 | [TK]D-Fender | :make a connection" doesn't mean almost anything by its raw wording |
23:19.55 | onculayin | Well I can make calls with internal extensions and make calls to external numbers, it works but just bad audio |
23:20.17 | onculayin | I can hear them very well, they have harder time hearing me (on the Android side) |
23:20.38 | onculayin | me (the Android client) |
23:20.38 | [TK]D-Fender | Levl is the app+device |
23:20.42 | [TK]D-Fender | quality = network |
23:21.23 | onculayin | Ok, but why can I hear them crystal clear but not them? IS that a sight for bad mic? |
23:21.26 | onculayin | sign |
23:21.37 | onculayin | Would a bad network effect both directions? |
23:21.59 | [TK]D-Fender | Could be the JB+PLC isn't the same grade on each end. |
23:22.09 | [TK]D-Fender | And could mean that maybe packets go one way better |
23:22.59 | onculayin | yeah that is what I am trying to figure out I guess. I can hear them very well no matter which wireless conection I am on. But that is not the case for the other end |
23:23.28 | onculayin | cosistently bad audio from my side to them |
23:23.42 | [TK]D-Fender | could be mic gain, poor AEC, etc |
23:23.53 | onculayin | I also tried the "Ring" app which seemed to provide the best audio |
23:24.15 | onculayin | [TK]D-Fender: thanks for the info, I will work on the mic side to see If I can improve it |
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23:32.42 | klow | pjsip driver - if I have a currently ongoing call with no audio on either end, is there some commands I can issue to see what is going on with the RTP? |
23:33.08 | klow | having no-way and 1-way audio calls between the same endpoints sproradically with no environmental or configuration changes and trying to sort out why |
23:37.34 | [TK]D-Fender | Clearly something changed in settings or environment. |
23:37.43 | [TK]D-Fender | otherwise nothing would have changed |
23:37.51 | [TK]D-Fender | rules make things go where they go |
23:37.58 | [TK]D-Fender | So you start with looking at the SIP debug |
23:38.29 | [TK]D-Fender | And prove where things really come fom, where they say they are coming from, where they say they are going, and where they say they should have gone |
23:42.57 | klow | Sorry to be clear what I mean is that I haven't moved from my location and IPs have not changed. Its possible my firewall is acting different from call to call and I assume that is the case |
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23:43.16 | klow | but I'm sitting here making the same call 3 times in a row and sometimes I get audio and sometimes not |
23:43.57 | klow | I see with "rtp set debug on" a bunch of data when the audio works but no messages when there is no audio |
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23:57.23 | *** part/#asterisk kharwell (kharwell@nat/digium/x-gewjupxjigjfvzar) |