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Asterisk No Audio in Extension - Stack Overflow
https://stackoverflow.com/questions/15800663/asterisk-no-audio-in-extension
Asterisk No Audio in Extension. Ask Question Asked 8 years, 9 months ago. Active 8 years, 4 months ago. Viewed ... the negotiated RTP port is probably being closed before RTP is sent between Asterisk and the phone. – Matt Jordan. Apr 4 '13 at 14:04. 1. I added progressinband=yes to my sip.conf general area and in now working. Thanks ...
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 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 …
[SOLVED] Elastix no audio between extensions - Asterisk ...
https://community.spiceworks.com/topic/964344-elastix-no-audio-between-extensions
Elastix no audio between extensions. Get answers from your peers along with millions of IT pros who visit Spiceworks. I installed a new elastix server , and created a organisation . Then made 2 extemsions . They connect perfectly from 2 different softphones from 2 PC but there is no audio between them .
[SOLVED] Remote extensions no audio - Asterisk PBX ...
https://community.spiceworks.com/topic/1052123-remote-extensions-no-audio
I am able get registration with my remote extensions, but no audio. I have tried many port forwarding options from Google, but still no audio. The extensions do produce a ring, but once answered no audio for the voice. I have tried every possible configuration not sure what else there is. Any ideas why there is registration and no audio?
No Audio between two internal extensions - FreePBX ...
https://community.freepbx.org/t/no-audio-between-two-internal-extensions/58199
So we’re using a PBXact 40 and two Sangoma phones internally for testing and for some reason there’s no audio between the two phones. They’ll ring each other but neither end can be heard. If I call voicemail for example though I hear audio from the system. Everythings configured as internal
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
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 hear one each other, I used from-internal context for both of them.
No audio between some extensions (strange) - Endpoints ...
https://community.freepbx.org/t/no-audio-between-some-extensions-strange/63653
The version we are using is FreePBX 14.0.1alpha34. everything was normal for years and then since last week i have started receive complains for no audio. here is the detail user told me. when user A calls user B both hear no audio. When the same user A calls user C audio is working fine. and when user C calls user A and user B audio is working ...
Remote SIP, no audio when using asterisk - Asterisk
https://forums.whirlpool.net.au/archive/925040
posted 2008-Feb-26, 8:07 am AEST. The general problem with the no audio on remote extensions with asterisk is where you put the sip nat settings. Most people put it in the sip_nat.conf file but as these settings would be added at the very end of …
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
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