Asterisk Locally Bridging Sip No Audio

We have collected the most relevant information on Asterisk Locally Bridging Sip No Audio. Open the URLs, which are collected below, and you will find all the info you are interested in.

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, ... on /etc/asterisk/sip.conf make sure you set the network address for the phones. You can alos add multiple networks, for example: localnet=172.16.1.0/24 localnet=192.168.1.0/24 ... Testing Asterisk SIP and DAHDI local calls. 0. Asterisk chan_mobile ...

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. ... – Locally bridging SIP/VOIPTEST-000000b8 and SIP/GO_VoIP_1-000000b9 ... (5060 and rtp port) and still ringing but no audio. Also i found that in sip show registry no host ...

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 …

SIP with NAT or Firewalls - Asterisk Guru

http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html

SIP with NAT or Firewalls. 1.1. Problem Description: Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange ...

Now you know Asterisk Locally Bridging Sip No Audio