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No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
2) Not answered calls: in your dialplan (/etc/asterisk/extensions.conf) you need to answer calls before they can send any audio (RTP), make sure every call runs through an Answer(). 3) Multiple answers in a single call: a call can be answered only a single time, in some asterisk versions you won't receive audio if a call is answered twice or more times so make sure you don't.
Calling using Google - Asterisk Project - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
Google lists supported audio codecs on this page - https://developers.google.com/talk/open_communications. Per section, 5, the supported codecs are: PCMA; PCMU; G.722; GSM; iLBC; Speex; Our experience shows this not to be the case. Rather, the codecs, supported by Asterisk, and seen in an invite from Google Chat are: PCMA; PCMU; …
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
All the other configuration options are not external to Asterisk. This is where I went wrong. I tried tweaking and changing asterisk settings for months to no avail. If you have No Audio - It's to do with your ROUTER or FIREWALL. If you've configured the ports as I've shown above, there are only a few things left to check. 1. Internal firewall.
Asterisk - sip phone has no sound - Stack Overflow
https://stackoverflow.com/questions/39006213/asterisk-sip-phone-has-no-sound
I have an asterisk server v 11.7 on an aws ec2 ubuntu 14.04 image but can't get any sound from a sip phone (either zoiper or linphone) over OpenVPN. I have tried both use DTMF SIP INFO and RFC2833 but neither works. The phones both play their default sounds ok. The asterisk server answers ok and extensions.conf is working fine in the CLI.
Asterisk and google talk - CentOS
https://forums.centos.org/viewtopic.php?t=15890
Asterisk and google talk. Post by cameron365 » Fri Jul 02, 2010 5:28 pm A few days i am googling to find the problem that killing me, i follow a lot of instruction to sovle my problem but it is not working. I have CentOS 5.5 and Asterisk 1.4.33.1 with res_jabber.so and chan_gtalk.so loaded.
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