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No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
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. Anyways, why is Asterisk placing 2 calls?
No audio for sip calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-for-sip-calls/71048
No audio for sip calls Asterisk Asterisk SIP AdamPetersen June 14, 2017, 6:26pm #1 So our asterisk is on an Azure server and we can register the sip phone, but don’t hear audio when making calls. We opened ports 15000-19000 on the firewall and changed rtp.conf to start at 15000 and end at 19000.
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
On your router, open up the following ports and forward them to your asterisk box. Your asterisk box may be in the form of (FreePBX, AsteriskNOW, or TrixBox)- They all use the same format SIP One Way Audio Troubleshooting Once these ports have been forwarded to the IP of your Asterisk server, give your router a reboot.
No audio - Fresh install - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-fresh-install/83257
Hello everyone, I’m fairly new to Asterisk and have a problem regarding audio. I have recently set up a VPS with CentOS 7 and Asterisk 16.9.0. It’s a basic, fresh install I did by compiling Asterisk myself. I used the instructions from the book “Asterisk: The Definitive Guide (Fifth Edition)” to setup everything. Endpoints are setup using a realtime connection with a …
Calls with no audio in both sides - Asterisk SIP ...
https://community.asterisk.org/t/calls-with-no-audio-in-both-sides/73244
The Mobile is associated with a Mobile Operator and has a Public IP address. Public IP access to the Asterisk Sagres server is by NAT filtering through a Firewall. That is, it a... Calls with no audio in both sides Asterisk Asterisk SIP humber2 January 19, 2018, 11:02am #1
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
I have installed magnusbilling on asterisk 11 and sometime no one of both end hear something, sometime I issued one way audio and sometimes I’m able to talk normally. In this third case i can see the output in “rtp set debug on”, otherwise no. ... – SIP/GO_VoIP_1-000000b9 answered SIP/VOIPTEST-000000b8 Audio is at 33636 Adding codec ...
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Without STUN support, you will also need NAT=yes or NAT=route, and you will have no incoming audio on the natted phone until the asterisk server received audio from that natted phone. 1.4.2 Asterisk as a SIP server outside nat, clients / proxies on the outside connecting to Asterisk There is no nat in between => no problem
PJSIP no audio on calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
Everything worked perfectly on chansip. I moved to PJSIP and I can’t hear audio on any of my calls. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. It worked well and I used the python script to convert sip.conf to pjsip.conf. If I enable direct media, I’m able to hear one-way. PJSIP config: removed RTP debug shows no log, nothing.
Asterisk and JsSIP no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-and-jssip-no-audio/84312
Asterisk and JsSIP no audio Asterisk Asterisk WebRTC vieridipaola May 28, 2020, 8:47am #1 Hi, I’m connecting a webrtc client to Asterisk 16, but I can’t hear the audio playback (dialed 200, and I should hear the demo-congrats audio file). Connecting with a SIP softphone works fine. The connections are within the same LAN segment.
webrtc - sipjs and asterisk voice call no audio issue ...
https://stackoverflow.com/questions/40396973/sipjs-and-asterisk-voice-call-no-audio-issue
I am giving asterisk log of both sip messages and rtp packet bellow: My problem is from log I am seeing that RTP packet is getting sent to both end via ICE. but in client browser no audio is playing. i.e. there is no audio in browser. <--- SIP read from WS:192.168.40.48:10380 ---> INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/WS 192.0 ...
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