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Asterisk - sip phone has no sound - Stack Overflow
https://stackoverflow.com/questions/39006213/asterisk-sip-phone-has-no-sound
I have an asterisk server v 11.7 on an aws ec2 ubuntu 14.04 image but can't get any sound from a sip phone (either zoiper or linphone) over OpenVPN. I have tried both use DTMF SIP INFO and RFC2833 but neither works. The phones both play their default sounds ok. The asterisk server answers ok and extensions.conf is working fine in the CLI.
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 …
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.
PJSIP no audio on calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
Everything worked perfectly on chansip. I moved to PJSIP and I can’t hear audio on any of my calls. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. It worked well and I used the python script to convert sip.conf to pjsip.conf. If I enable direct media, I’m able to hear one-way. PJSIP config: removed RTP debug shows no log, nothing.
Asterisk 15 on AWS, Correctly configured NAT, No audio on ...
https://community.asterisk.org/t/asterisk-15-on-aws-correctly-configured-nat-no-audio-on-sipml5-based-soft-phone/75459
Hi all, I am not able to hear any audio on my sipML5 based sip phone. Not able to send audio to asterisk either. (one-way audio issues) My stack: I have my asterisk (15.x version) deployed on AWS cloud, with all the ports open (since we are in the development phase).
Asterisk 11 for NAT. No Sound One Side, No Sound \ Voice ...
https://voipwifiphones.com/asterisk-11-for-nat-no-sound-one-side-no-sound-voice-how-to-solve/
Overcoming NAT for Asterisk can be very difficult (there is no sound) because RTP traffic and SIP signaling go separately. On the Internet, almost all descriptions of the NAT option settings are reduced to the older version of Asterisk 1.8. Let's try to consider the settings options for the current Asterisk 11 -
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
No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
But when I am trying to do the same between two mobile numbers (through my sip provider), I get all the signals (ringing, hangup) on both phones, but no audio on both phone. Sample PJSIP config for my sip provider I am using as: [sipprovider] type=endpoint transport=transport-udp aors=sipprovider auth=sipprovider outbound_auth=sipprovider ...
SIP Registering. Rings External Phone. No Audio/Sound ...
https://community.freepbx.org/t/sip-registering-rings-external-phone-no-audio-sound/31325
SIP Registering. Rings External Phone. No Audio/Sound FreePBX 12.0.76.2 Asterisk 11.19.0 Linux 2.6.32-431.el6.i686 I am using SIP Station trunks. The sips are registering. I am able to ring my cell phone from the handset But there is no audio from the handset The SIP trunks are in use on a second server and working fine.
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