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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 available on SIP - Asterisk Support - Asterisk ...
https://community.asterisk.org/t/no-audio-available-on-sip/69589
app.c:1242 __ast_play_and_record: No audio available on SIP/ I am using codec as ulaw and alaw. This is my sip configuration : type=friend dtmfmode=rfc2833 qualify=yes insecure=port,invite nat=force_rport,comedia username=***** secret=***** host=XX.XX.XX.XXX port=5060 context=inbound disallow=all allow=alaw allow=ulaw. Extension as below :
Problem with voicemail "No audio available on SIP/MY NU ...
https://community.asterisk.org/t/problem-with-voicemail-no-audio-available-on-sip-my-nu/11493
Hi all! I recently installed asterisk v. 1.4 after reinstalling my server with fedora core 5 (from FC 3 with asterisk 1.2). Everything worked fine with asterisk installation and I copied over my .conf files from old asterisk version. The system is running fine - I can receive and make calls via my IP-phone provider which is registered in sip.conf. The problem with this new verision of …
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.
play_and_record: No audio available - asterisk-users@lists ...
https://asterisk-users.digium.narkive.com/o6n9cHEM/play-and-record-no-audio-available
When trying to leave a voicemail on one of the extensions, the Asterisk console reports: Feb 7 23:26:17 WARNING[409618]: app_voicemail.c:1200 play_and_record: No audio available on SIP/2001-dc12?? and the resulting .wav file in /var/spool/asterisk is empty/invalid. I did find a thread (via google) that described someone with the same
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 ...
Nat aws ec2 pjsip - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/nat-aws-ec2-pjsip/91541
Hi guys i have a problem with my pjsip asterisk configuration hosted on aws ec2. My phones are not ringing and i have no audio. Thx for help, my config : [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0 ;local_net=xxx.xxx.x.x/24 local_net=xxx.xx.x.xxx/20 ; my local ip aws external_media_address=x.xxx.xx.xxx ; my public ip aws …
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