Asterisk H323 No Audio | Audio-Digital.net

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[HELP] SIP -> H323 No Sound Problem - Asterisk Support ...

https://community.asterisk.org/t/help-sip-h323-no-sound-problem/4881

Asterisk CLI/SIP Debug/OOH323 Debug never indicate a problem. I have found a couple of helpful things here: Using the “Dial” command from the CLI, outbound SIP calls work fine and outbound H323 calls fail, indicating this is an H323 problem, not SIP. The second thing is to enable H323 logging. I will post the log file at the end of this post.

H.323 problem - Asterisk Support - Asterisk Community

https://community.asterisk.org/t/h-323-problem/7337

I am using chan_h323 to connect to h.323 sources. I am using Netmeeting to generate h.323 traffic. When I attempt to call extension 300, I hear no audio coming back from Aterisk, even though it is being sent. Here are various config files and Atersik CLI output. h323.conf ; The NuFone Network’s ; Open H.323 driver configuration ; [general] port = 1720 …

Ooh323<->pjsip audio call - Asterisk Support - Asterisk ...

https://community.asterisk.org/t/ooh323-pjsip-audio-call/89426

I install GNUGK and Asterisk on one cloud server with a different port. There 2 calls succeeded, and 2 calls failed, all audio calls. yate h323 softphone (support faststart) => gnugk => asterisk.ooh323 (faststart=yes) => asterisk.pjsip => microsip softphone (this h323 → sip call succeed)

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 ...

One-way audio or not audio at all - Kolmisoft Wiki

http://wiki.kolmisoft.com/index.php/One-way_audio_or_not_audio_at_all

Make canreinvite = no in device settings Device is H323 . If while connecting with an H.323 client, you get no audio or garbled audio and messages like this on the Asterisk console: rtp.c:489 ast_rtp_read: Unknown RTP codec 107 received, try to disable Speex or some other codec on Asterisk and/or the client side.

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