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No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/65068#:~:text=If%20you%20still%20don%E2%80%99t%20get%20audio%2C%20report%20whether,give%20you%20audio%2C%20add%20this%20to%20%2Fetc%2Fasterisk%2Fpjsip_custom_post.conf%20disable_direct_media_on_nat%3Dyes
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 ...
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 …
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 Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
No sound with SIP | No Sound With Asterisk| No Sound With NAT. To correctly troubleshoot your issue, make sure that the following steps are taken. 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.
Asterisk 16.4 & WebRTC = no audio - Asterisk WebRTC ...
https://community.asterisk.org/t/asterisk-16-4-webrtc-no-audio/83624
Hello there, we’ve just tried our fist webrtc test with Asterisk 16.4.0 The endpoints get registered well and it’s able to send calls without problems … but without audio. This is the Asterisk CLI: -- Executing […
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
Hi all, I’m new in voip and asterisk and i really need your help. 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. What can be the problem? What do you need for …
Calls with no audio in both sides - Asterisk SIP ...
https://community.asterisk.org/t/calls-with-no-audio-in-both-sides/73244
I’m trying to implement a new service on my Asterisk server network. We have a 3G mobile phone with a softphone app - Linphone, whose number is 9012 and is registered on the Asterisk server with IP # 10.192.124.101 (Sagres). Then there is a Trunksip between this server and another one with IP # 10.192.230.231 (Viriato). The Mobile is associated with a Mobile …
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Make sure asterisk sends the messages faster than the timeout on your NAT device. 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.
No audio on inbound or outbound calls - General Help ...
https://community.freepbx.org/t/no-audio-on-inbound-or-outbound-calls/47303
I am running FreePBX with Asterisk version 15.1.5. I am using an old ObiHai 110 device as an FXO port and a Gigaset C530IP DEC station as a PJSIP extension. Both devices register with PBX and calls can be made and received but there is no audio in either direction. When calling the extension’s voicemail, the logs show that the proper audio files are played by …
No audio on softphones, but rings - Endpoints - FreePBX ...
https://community.freepbx.org/t/no-audio-on-softphones-but-rings/65494
Here is the updated link with the SIP debug turned on. Called from the deskphone (ext 105) to the softphone (ext 205) Got ring, got successful connect, but no audio. And as near as we can tell neither end can hear the other speak. The log starts too late (with the ACK to the INVITE we are interested in).
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