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| 01:10.45 | jfindley | hello, i'm acting as a B2BUA for a customer who wants to send 183 followed by a disclaimer to the caller, but asterisk isn't passing the audio on to the caller. I've tried all the combinations of prematuremedia and progressinband but asterisk won't pass the early media on to the caller. |
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| 02:55.55 | snadge | if i have something like: |
| 02:56.09 | snadge | exten => _X.,n,ExecIF($[${CALLERID(num)}=anonymous]?Set(CALLERID(num)=private)) |
| 02:56.22 | snadge | but i also want to do that if CALLERID(num) is null.. |
| 02:57.48 | snadge | so i could maybe put something like ISNULL(${CALLERID(num)}) .. or just delete the anonymous.. i dont know |
| 03:00.12 | radicaldev | yep, add an OR in the if test |
| 03:00.58 | snadge | syntax for that? :| |
| 03:01.27 | radicaldev | verticle pipe: https://wiki.asterisk.org/wiki/display/AST/Operators |
| 03:01.43 | radicaldev | (cid=anonymous | isnull(cid)) |
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| 03:02.39 | snadge | the sad thing is im a programmer.. but.. wrapping my head around the dialplan syntax is doing me in |
| 03:02.50 | snadge | i didnt write that above code.. but why is there a $[ |
| 03:03.51 | snadge | that must be similar to a bash style test.. eg.. if whats between [] is true, then set cid private |
| 03:04.52 | radicaldev | Yeah, I'm not sure what the guys who did the dialplan syntax were on about, but once you get used to it it's pretty expressive |
| 03:05.27 | radicaldev | do you understand anything about that command? |
| 03:07.00 | radicaldev | that line reads more or less like: when the extension is _X and we're at priority n, then set the cid to private if the cid is anonymous |
| 03:07.32 | radicaldev | you want to add 'or empty' to the end of it |
| 03:09.13 | radicaldev | that looks like: exten => _X,n,ExecIF($[${CALLERID(num)}=anonymous | isNull(${CALLERID(num)})]?Set(CALLERID(num)=private)) |
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| 03:21.32 | snadge | yeah i managed to get it sorted .. thats what i did yeah |
| 03:21.37 | snadge | thanks :) |
| 03:21.45 | radicaldev | no worries |
| 03:21.47 | snadge | but.. the problem is somewhere else of course.. hehe |
| 03:22.04 | radicaldev | what's the troiuble now? |
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| 11:53.31 | pa | to enable tcp AND udp in asterisk sip, is it enough to have tcpenable=yes? |
| 12:12.37 | mirela666 | pa, i think it will be udp by default and with tcpenabe will go over tcp only |
| 12:12.44 | mirela666 | not 100% sure |
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| 14:04.11 | gingitsune | If I'm starting out with asterisk and I want to integrate asterisk in our crm, how much telephony do i need to know? |
| 14:04.29 | gingitsune | And do you have a nice resource that covers the essentials? |
| 14:05.03 | Penguin | How much telephony do you want to integrate into your CRM? |
| 14:05.42 | gingitsune | Just handling the calls, the actual telephony part is still handled with hardware phones |
| 14:05.57 | gingitsune | Ansering, displaying queues, transfering etc |
| 14:07.23 | gingitsune | answering* |
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| 14:18.39 | gingitsune | i have to go, ill be back in a while |
| 14:22.42 | [TK]D-Fender | ~book |
| 14:22.42 | 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 |
| 14:22.44 | [TK]D-Fender | ^^^ |
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| 15:18.50 | pa | mirela666, thanks |
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| 15:19.37 | pa | so i also tried to add transport=tcp, and the client registers fine with the server |
| 15:19.57 | pa | however, when i place a call, i hear nothing and i see transmitted bytes many but received bytes = 0 |
| 15:19.59 | pa | not sure why |
| 15:20.13 | pa | if i need another port |
| 15:20.18 | pa | and if that port has to be udp |
| 15:20.57 | pa | or if it's about RTP being blocked |
| 15:23.13 | pa | i guess i need rtp over tcp somehow |
| 15:23.15 | pa | tunnel or something |
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| 15:25.06 | gingitsune | Hey, I'm back |
| 15:25.41 | gingitsune | I had a question: how much do i need to know about telephony to integrate asterisk into a crm |
| 15:26.01 | gingitsune | We need just basic things like making, droping calls |
| 15:26.07 | gingitsune | Redirecting them etc |
| 15:26.21 | gingitsune | And can anyone recommend a soft introduction on the topic |
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| 15:32.42 | [TK]D-Fender | ~book |
| 15:32.43 | 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 |
| 15:32.45 | [TK]D-Fender | ^^^ |
| 15:33.45 | [TK]D-Fender | the basics for voip itself isn't all thatt much. More complicated will be the dialplan and other related bits |
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| 15:53.21 | gingitsune | Thanks! |
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| 17:19.55 | flujan | hello, is there a func_odbc equivalent to postgresql in asterisk? |
| 17:21.47 | [TK]D-Fender | ? |
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| 17:22.55 | flujan | [TK]D-Fender: not using odbc a native one. :) |
| 17:23.13 | [TK]D-Fender | The smallest search will tell you "no" |
| 17:23.21 | [TK]D-Fender | no native |
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| 18:12.34 | ledoktre | hey y'all. Im wondering if anyone here knows what command asterisk uses to combine teh two channels (monitor) to stereo when you use the 'm' option. Im trying to do it in a script and having trouble figuring out how to get sox to read 2 wav49 files and mux them together. |
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| 18:43.42 | ledoktre | bump |
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| 19:08.40 | jfindley | https://wiki.asterisk.org/wiki/display/AST/Application_Monitor |
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| 19:09.17 | jfindley | MONITOR_EXEC (default is sox -M in.wav out.wav something.wav) |
| 19:13.31 | ledoktre | jfindley: I am seeing something like that in the source code, but Ive tried it, and it isn't putting them together that I can tell. It just writes one of the files to the wav. Im trying to use wav49 - not standard wav. When I use standard wav it works fine. I just want it smaller for emailing |
| 19:14.42 | WIMPy | You can't output stereo to a mono format, obviousely. |
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| 19:55.55 | jfindley | WIMPy: I think he's trying to merge two mono files into a mono file. Totally doable. |
| 19:56.47 | jfindley | I need some help figuring out how to pass early media to the caller from a downstream provider so they can play a disclaimer |
| 19:57.14 | jfindley | i get the 183/SDP followed by RTP but asterisk doesn't send it on |
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| 20:01.27 | WIMPy | did read the word stereo. |
| 20:08.04 | ledoktre | i had wondered about that. i had googled on it but had not found much on it. i did a test just now and it does appear it joins them. It threw me - same file size. But the files are muxed together. Thanks. |
| 20:25.10 | Freenex | jfindley may progressinband parameter help you? |
| 20:25.43 | Freenex | I didn't try it but I read this http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband |
| 20:36.13 | jfindley | I tried working with that, set to yes, no, never.. same result. they stream the RTP to me from the called side and the RTP stops at asterisk |
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| 20:51.05 | Freenex | Maybe prematuremedia parameter in combination with progressinband? It sems asterisk process Early media to deal with some ISDN providers |
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| 21:24.08 | jfindley | I tried PI=yes, PM=no - PI = no, PM =no - PI=no, PM=yes - PI=yes, PM=yes ---- and it doesn't pass the audio through |
| 21:24.20 | jfindley | i'm thinking maybe it's a dial command option or something I'm looking for |
| 21:51.29 | igcewieling | jfindley: comment out both settings and put Progress() in your dialplan. See if that helps. |
| 21:53.11 | jfindley | Progress before dial? or just somewhere early on? |
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| 23:28.21 | wyoung | jfindley: did you try PI=3.141592653589793 ? |
| 23:28.39 | jfindley | Nope |
| 23:29.07 | wyoung | ok well don't :) |
| 23:29.14 | wyoung | I got bored |
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