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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. ... m=audio 44448 RTP/AVP 107 119 100 106 0 105 98 8 101 a=alt:1 2 : eQpofyBA 4H3455IQ 172.31.80 ...
Wrong SDP details, RTP and NO Audio - Asterisk SIP ...
https://community.asterisk.org/t/wrong-sdp-details-rtp-and-no-audio/77464
Hello Team, As a carrier we have a customer using Asterisk version 13, and we authenticating the SIP registration over an IP. [ Trunk is Registered ] On her INVITES header, here is what I noticed, v=0 o=root 1995318922 1995318922 IN IP4 192.168.3.17 s=Asterisk PBX 13.16.0-rc2 c=IN IP4 41.58.129.215 t=0 0 m=audio 11300 RTP/AVP 8 0 101 a=rtpmap:8 …
Asterisk gives "Strict RTP learning" message and no …
https://stackoverflow.com/questions/49682747/asterisk-gives-strict-rtp-learning-message-and-no-audio-for-chrome-webrtc-but
Asterisk gives "Strict RTP learning" message and no audio for Chrome WebRTC but works in Firefox. Ask Question Asked 3 years, 8 ... (works on Firefox, connects but has no sound on Chrome) with the same debug output in Asterisk except that the initial address is 0.0.0.0:9 instead of 127.0.0.1:9. Regardless I'm not sure what next steps to even ...
PJSIP, NAT and RTP: no audio - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/pjsip-nat-and-rtp-no-audio/72915
Basically, RTP is going to the endpoint local IP address, so I don’t have audio. Here’s the relevant configs: Local server IP: 192.168.44.6 WAN server IP: 131.161.42.186 Local endpoint IP: 192.168.42.10 WAN endpoint…
No audio for sip calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-for-sip-calls/71048
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. Here is the output I get when calling an extension that is supposed to just hangup immediately. I changed the IP of the computer with the soft phone …
Asterisk 16.4 & WebRTC = no audio - Asterisk WebRTC ...
https://community.asterisk.org/t/asterisk-16-4-webrtc-no-audio/83624
[2020-04-16 13:54:28] VERBOSE[32020][C-00000267] res_rtp_asterisk.c: Sent RTP packet to 88.5.35.95:64260 (type 111, seq 007662, ts 000960, len 000070) [2020-04-16 13:54:28] VERBOSE[32020][C-00000267] res_rtp_asterisk.c: Sent RTP packet to 88.5.35.95:64260 (type 111, seq 007663, ts 001920, len 000069) [2020-04-16 13:54:28] VERBOSE[32020][C ...
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. ... but there is no RTP packet that is sent by the Sagres server. ... m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102 a ...
No audio on PJSIP channels - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/no-audio-on-pjsip-channels/90281
If Asterisk is behind NAT and you are using ICE, then you have to configure the ice_host_candidates portion of rtp.conf with a mapping for external IP address or enable STUN so Asterisk can discover its external IP address otherwise ICE negotiation will fail, and the call will have no media provided the client is external.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
(STUN would not have to send RTP to your asterisk server to make the binding, only something to the STUN server). 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 …
Remote SIP, no audio when using asterisk - Asterisk
https://forums.whirlpool.net.au/archive/925040
There was no RTP stream to be seen, so I filtered out the initial session and looked at the asterisk to VSP session and noted it was connected to asterisk RTP port, so I forwarded this on my router behind which asterisk sits and that fixed the issue. ... The general problem with the no audio on remote extensions with asterisk is where you put ...
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