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No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
Hi @grvtslyr, maybe it helps to see the logs what actually is done by asterisk. I’m using rasterisk -vvvvv to see the dialplan and other details like RTP connections, for example:. 0x7fee80096ac0 – Strict RTP learning after remote address set to: 192.168.1.14:12540. This may be easier to track down your issue.
Asterisk webrtc no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-webrtc-no-audio/91333
I config asterisk for webrtc and sipjs.com. so that work for SimpleUser configuration. with SimpleUser every thing is Ok. but when I use UserAgent , connection has established without audio .this is asterisk console when User3 call User2 : WebSocket connection from '2.185.145.48:45486' for protocol 'sip' accepted using version '13' -- Added contact …
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...
Two extensions on the same LAN but no audio - Asterisk SIP ...
https://community.asterisk.org/t/two-extensions-on-the-same-lan-but-no-audio/74360
My asterisk is on a remote site behind an NAT router. I have two extensions 3001 (192.168.7.32) and 3008 (192.168.7.93) on the same LAN in a separate site with external IP 176.156.223.236. These two extensions are registered to the asterisk. My network works pretty well. These two extensions can call in and out, including via a SPA 3000 to the POTS world. …
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
SIP One Way Audio Troubleshooting. Once these ports have been forwarded to the IP of your Asterisk server, give your router a reboot. Re-Login to your router just to reconfirm that the posts are in fact pointing to the correct IP of the Asterisk Server. The next step is to ensure that you configure your NAT settings on the Asterisk server correctly.
WEBRTC no audio - Asterisk WebRTC - Asterisk Community
https://community.asterisk.org/t/webrtc-no-audio/90896
WEBRTC no audio. Asterisk Asterisk WebRTC. bobtarask December 11, 2021, 6:19pm #1. Hello there ! I’m trying to use WEBRTC with SIPJS and Asterisk (Asterisk GIT-16-89cf7899be) . It works very well with Chrome version ~50. Since I’m using chrome 96 on Android 10 audio seems not working at all…. Maybe codecs change?
No audio on PJSIP channels - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/no-audio-on-pjsip-channels/90281
Hello, I don’t understand why I have no audio on my endpoints, When I log my softphones from a local address, everything works fine, but if I log from outside my network I have no audio. I don’t understand how Astertisk can match the local IP address of my client, since In my softphone code I use my public IP address to log to asterisk I’m using WebRTC for my …
No audio on sip calls over VPN - Endpoints - FreePBX ...
https://community.freepbx.org/t/no-audio-on-sip-calls-over-vpn/78414
No audio on sip calls over VPN. i have a FreePBX (asterisk) system as my pbx. It is connected to my Mikrotik. I have two Mikrotik i have setup server l2tp VPN and client VPN. Inside my internal lan, 10.0.0.0/24, everything is working fine as voip telephony concerned. When i connected through VPN, i can register my sip phone and i can call every ...
No audio on remote extension - Configuration - FreePBX ...
https://community.freepbx.org/t/no-audio-on-remote-extension/71988
I setup the bridge but that did not make any difference to my no audio problem. Later I saw Lorne’s question and went to double-check my settings. That is when I noticed and added the remote address under the pj_sip tab and did a restart (core restart now using the CLI) that is when things began to work.
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