IRC log for #asterisk on 20160226

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00:39.01F2KnightPer [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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01:00.09F2Knight[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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10:12.29stefan27how 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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11:04.03mohan__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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18:07.42lvlinuxAsterisk 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-Fendercorrect
18:11.55lvlinuxWhat 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-Fendera few things may just not make sound, others = hard drop
18:12.28[TK]D-Fendermost = drop
18:13.17lvlinuxk thanks. I think we'll just continue to avoid g729 then :-)
18:14.44lvlinuxI 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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18:53.04txt23Hey 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-FenderIt's a SIP requuest for a call processing server.
18:54.23[TK]D-FenderWhich you can forget with *
19:01.18txt23So I used sip:music@iptel.org and it worked. :)
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19:03.24[TK]D-FenderIf they host openly like that and you want them trying to reinvite outt, then you can try that...
19:03.41[TK]D-Fenderbut it leaves your network and I'm not sure how it'll react if it fails
19:03.55[TK]D-FenderAnd 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
19:05.00txt23Yeah 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
19:05.49infobotAsterisk: 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-FenderAnd read the sample config for it
19:07.33Tachohwe 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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23:02.20onculayinHow 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-FenderSoftware generally isn't going to make things better
23:13.13[TK]D-Fenderbad audio is either a bad mic, or bad network
23:13.38[TK]D-Fenderjitter + PL = bad
23:17.25onculayin[TK]D-Fender: thanks for the info. Would buying the codec from Asterisk help?
23:18.08[TK]D-FenderDepends if you're choking on BW.
23:18.22[TK]D-FenderBut if that's your issue... the tiny savings is unlikely to save you anything
23:18.27[TK]D-Fenderagain... NOT software
23:18.52[TK]D-Fenderunless one does slightly better PLC than others.
23:18.59onculayin[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-Fenderbut 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
23:19.55onculayinWell I can make calls with internal extensions and make calls to external numbers, it works but just bad audio
23:20.17onculayinI can hear them very well, they have harder time hearing me (on the Android side)
23:20.38onculayinme (the Android client)
23:20.38[TK]D-FenderLevl is the app+device
23:20.42[TK]D-Fenderquality = network
23:21.23onculayinOk, but why can I hear them crystal clear but not them? IS that a sight for bad mic?
23:21.26onculayinsign
23:21.37onculayinWould a bad network effect both directions?
23:21.59[TK]D-FenderCould be the JB+PLC isn't the same grade on each end.
23:22.09[TK]D-FenderAnd could mean that maybe packets go one way better
23:22.59onculayinyeah 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
23:23.28onculayincosistently bad audio from my side to them
23:23.42[TK]D-Fendercould be mic gain, poor AEC, etc
23:23.53onculayinI also tried the "Ring" app which seemed to provide the best audio
23:24.15onculayin[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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23:32.42klowpjsip 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?
23:33.08klowhaving 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-FenderClearly something changed in settings or environment.
23:37.43[TK]D-Fenderotherwise nothing would have changed
23:37.51[TK]D-Fenderrules make things go where they go
23:37.58[TK]D-FenderSo you start with looking at the SIP debug
23:38.29[TK]D-FenderAnd 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
23:42.57klowSorry 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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23:43.16klowbut I'm sitting here making the same call 3 times in a row and sometimes I get audio and sometimes not
23:43.57klowI 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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