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Getting Asterisk to Bridge Audio | MCB Systems
https://www.mcbsys.com/blog/2008/11/getting-asterisk-to-bridge-audio/
From Joshua Colp, Software Developer at Digium, comes this explanation of bridging: Packet2Packet Bridging = Audio is not going through the Asterisk core, it comes into the RTP stack and goes directly out. This decreases the amount of memory allocation that happens, and things require less processing.
No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
Hi all, I am facing one weird issue with asterisk. I am using manager api to originate calls. I am successfully able to do that between one direct channel (on webrtc) and mobile number (through my sip provider) while everything works perfectly (able to hear voice, can end the call etc). But when I am trying to do the same between two mobile numbers (through …
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 ...
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 …
Yealink T46G Problema al capturar llamada - Foros Asterisk ...
https://asteriskmx.org/foros/forum/asterisk/hardware-aa/38938-yealink-t46g-problema-al-capturar-llamada
-- Packet2Packet bridging SIP/6000-0053b418 and SIP/4001-0053b430 segunta prueba: telefono cisco a telefono cisco no solo el emisor de la llamada escucha audio, quien capturo (telefono yealink estension 4001 no escucha nada) == Spawn extension (administrativo, 3012, 1) exited non-zero on 'SIP/3010-0053ba76'
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
Audio issues - dropped packets - FreePBX Community Forums
https://community.freepbx.org/t/audio-issues-dropped-packets/54045
Audio issues - dropped packets. Bradbpw 2018-11-02 23:16:16 UTC #1. This is a continuance of a previous thread that was closed since a new reply wasn’t added within 7 days. - Help with network setup - audio breaks up. To summarize, I’m dropping packets and getting an audio break-up. Here is a sample of my RTP Debug.
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.
What is Locally Bridging? : Asterisk - reddit
https://www.reddit.com/r/Asterisk/comments/5936q6/what_is_locally_bridging/
Thanks to RTP and bridging rewrites, the terminology is a bit different now (since about Asterisk 10, I believe). Now "native" and "Packet2Packet" bridges are both considered to be "native" bridges. The old "native" bridge is called a remote native bridge. The old "Packet2Packet" bridge is called a local native bridge.
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