Before I got an log a ticket, can I just check Im not doing anything wrong?In 15.2, to install Opus:1) run `make menuselect`2) Highlight Codec Translators and press enter.3) Scroll down to codec_opus in the section labeled External4) Press enter to sel..
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile.But in version 13.1 (I think version 11.2 also) is not working l..
hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didnt work with pjsip in asterisk12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems ..
Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDIuse alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to al..
every one i want to have multiple sip calls with different codecs for each one. for example call to 8100 has g729 codec while call to 7900 has ulaw codec. i searched a lot and found that there is some variable like sip_codecwhich can set codec fo..
I use asterisk realtime, and I can set the order of codec preference on my realtime allow column.If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on C..
I have installed and configures this card in asterisk 1.6. When trying to load the module codec_sangoma.so I see the following in the asterisk log. [2012-06-04 15:50:31] WARNING loader.c: Error loading module codec_sangoma.so: /usr/lib/asterisk/modules/codec_sangoma…
Greetings List. I Have a small test server and im facing a small issue. i have setup two SIP PEERS and they are able to do Video calls. now Im testing SET SIP_CODECin a dial plan and when ever im setting the codec .. the inbound (=first) leg stops receiv..
we are working on a new codec for asterisk.Its early stages and the main goal is just to be able to hear it using normal client end points.So what I want is to be able to take a couple of normal G711 extensions, but have asterisk internally force a transc..
I have been breaking my head on this, cant find a solution. Anyone know a way to mute DTMF on SIP? I have already tried changing the dtmfmode option and messing with different codec/dtmfmode settings but so far, not having any luck. Not even sure chang..