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Solution to the Asterisk problem – no sound when calling ...
https://ixnfo.com/en/solution-to-the-asterisk-problem-no-sound-when-calling-via-nat.html
1. directmedia=no. Earlier in older versions of asterisk, instead of directmedia=no, canreinvite=no was used. To support a NAT connection, specify the qualify parameter: 1. 2. qualify=yes. ;qualify=300. Also in the “general” section you can manually specify the local network and the external asterisk IP address for connections, for example:
Calls with no audio in both sides - Asterisk SIP ...
https://community.asterisk.org/t/calls-with-no-audio-in-both-sides/73244
Calls with no audio in both sides. Asterisk. Asterisk SIP. humber2 January 19, 2018, ... canreinvite=no canreinvite=yes directmedia=yes qualify=yes nat=no deny=0.0.0.0/0.0.0.0 ... as the source address is not a known peer and allowguest=no! Asterisk seems to have sent its private network address as media address to the requestor. That’s ...
No sound on incoming calls - Asterisk Support - Asterisk ...
https://community.asterisk.org/t/no-sound-on-incoming-calls/24607
Here’s the strange problem. I’ve set up Asterisk to connect to Broadvoice as the trunk, and then I have a SIP phone here on my desk which connects to Asterisk. I’ve set up a dialplan on Asterisk so that my SIP phone here goes into a context called staff. From within that context, it can dial out through the Broadvoice SIP line. It works fine. I can dial out and get good …
Call established, no sound - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/call-established-no-sound/84867
Asterisk. Asterisk SIP. menticol. July 9, 2020, 8:41am #1. Hi guys! ... (shown as 6666666666) to the extension 12345, the established call has no audio. What I already tried: I enabled anonymous calls (before this the call could not even be established) ... canreinvite=no disallow=all echocancel=yes allow=ulaw callgroup=1.
[HELP]: No audio during first 10 seconds of SIP calls ...
https://community.asterisk.org/t/help-no-audio-during-first-10-seconds-of-sip-calls/33821
system January 13, 2011, 11:15pm #1. SIP to SIP calls connected through Asterisk only get audio about 10 seconds or so after the call connects. However, calls answered by the an application (namely voicemail) get audio as soon as the call connects AND calls setup from the command line ‘channel originate…’ also get audio right away. The forums have a lot of ‘no …
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