We have collected the most relevant information on Asterisk Sip Trunk No Audio. Open the URLs, which are collected below, and you will find all the info you are interested in.
Calls forwarded outbound via SIP trunks connect but no audio
https://www.tek-tips.com/viewthread.cfm?qid=1767037#:~:text=Forwarding%20a%20call%20coming%20into%20the%20SIP%20trunks,trunks%20to%20a%20SIP%20extension%20works%20just%20fine.
SIP Trunk, No NAT, One Way Audio - Asterisk Support ...
https://community.asterisk.org/t/sip-trunk-no-nat-one-way-audio/33583
Hi, Asterisk 1.6.2.11. SIP trunk from an operator. Outgoing call : signal is OK, audio is only one way. I can hear the distant person, but she can’t hear me. Networking : [IP Phone] ------- [Asterisk] ------- [VoIP Provider's Router] ------- [VoIP Provider's PABX] 192.168.4.107 eth0 : 192.168.4.1 eth1 : 37.1.1.1 37.1.1.2 212.39.140.250 pabx*CLI> sip show peers altitude-output …
No audio - DIDWW to Asterisk SIP Trunk - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-didww-to-asterisk-sip-trunk/68467
hi, I’ve a SIP trunk between DIDWW to Asterisk SIP Trunk. The call comes in to the Asterisk but there is no IVR sound. Asterisk Trunk Configuration context=from-trunk dtmfmode=rfc2833 dtmf=rfc2833 host=X.X.X.X insecure=very type=peer nat=never (I tried “yes” and “route” - nothing helps) allow=g729 (I tried ulaw, alaw & all - nothing helps) Please find the …
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 when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
No audio when bridging two trunk calls. Asterisk. Asterisk SIP. grvtslyr. September 17, 2020, 9:26am #1. Hi all, ... I get all the signals (ringing, hangup) on both phones, but no audio on both phone. Sample PJSIP config for my sip provider I am using as: [sipprovider] type=endpoint transport=transport-udp aors=sipprovider
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
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 …
No audio on outgoing trunk call - General Help - FreePBX ...
https://community.freepbx.org/t/no-audio-on-outgoing-trunk-call/56697
For some outbound trunk calls to specific external destinations, the internal caller can’t hear the remote side’s audio for about 10-20 seconds although the remote can hear the caller. My setup is a CentOS 6.10, FPBX-13.0.192.19, Asterisk 13.23.0. Here’s a link to the network diagram. Also here are some settings:
asterisk: IP address order may cause no audio · Issue …
https://github.com/irontec/ivozprovider/issues/511
The first SIP message, the invite from trunk provider to my system shows no difference at all. The second SIP message, the INVITE from kamailio trunks to asterisk has a difference on the last SDP header. The IPs: xxx.xxx.xxx.194 kamailio trunks xxx.xxx.xxx.195 kamailio users / rtp proxies audio sock xxx.xxx.xxx.206 trunk provider SIP server
Calls forwarded outbound via SIP trunks connect but no audio
https://www.tek-tips.com/viewthread.cfm?qid=1767037
Forwarding a call coming into the SIP trunks and then back out the SIP trunks results in no audio. The call rings on the far-end, and can be answered, but there's no audio in either direction. A call inbound via SIP trunks to a SIP extension works just fine. (Using a softphone on my laptop registered via VPN during testing)
No audio on softphones, but rings - Endpoints - FreePBX ...
https://community.freepbx.org/t/no-audio-on-softphones-but-rings/65494
If you still have trouble, at the Asterisk command prompt, type sip set debug peer xxx where xxx is the extension number. Make a failing test call, paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here. Also, report whether the desk phone user can hear the softphone user, and vice-versa.
Now you know Asterisk Sip Trunk No Audio
Now that you know Asterisk Sip Trunk No Audio, we suggest that you familiarize yourself with information on similar questions.
