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Asterisk 11 for NAT. No Sound One Side, No Sound \ Voice... How …
https://voipwifiphones.com/asterisk-11-for-nat-no-sound-one-side-no-sound-voice-how-to-solve/#:~:text=Overcoming%20NAT%20for%20Asterisk%20can%20be%20very%20difficult,for%20the%20current%20Asterisk%2011%20%E2%80%93%20Asterisk%20settings.
Solution to the Asterisk problem – no sound when calling ...
https://ixnfo.com/en/solution-to-the-asterisk-problem-no-sound-when-calling-via-nat.html
1. directmedia=no. Earlier in older versions of asterisk, instead of directmedia=no, canreinvite=no was used. To support a NAT connection, specify the qualify parameter: 1. 2. qualify=yes. ;qualify=300. Also in the “general” section you can manually specify the local network and the external asterisk IP address for connections, for example:
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 – Asterisk settings.
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
3) Multiple answers in a single call: a call can be answered only a single time, in some asterisk versions you won't receive audio if a call is answered twice or more times so make sure you don't. Anyways, why is Asterisk placing 2 calls?
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
From the above dial plan when I dial 100 it should play me hello-world, with zoiper mobile app on wifi(NAT) and mobile data(no NAT situation), I am able to connect to asterisk and hear proper audio and also I am seeing RTP packets communication between zoiper and asterisk.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Make sure asterisk sends the messages faster than the timeout on your NAT device. 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. 4) call coming from asterisk outside the nat with a Restricted Cone Nat device
PJSIP, NAT and RTP: no audio - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/pjsip-nat-and-rtp-no-audio/72915
PJSIP, NAT and RTP: no audio. Asterisk. Asterisk SIP. thiagocnet December 13, 2017, 12:13pm #1. Basically, RTP is going to the endpoint local IP address, so I don’t have audio. ... I’m doing a “originate” so only Asterisk sends audio to the endpoint.
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
If you have No Audio - It's to do with your ROUTER or FIREWALL. If you've configured the ports as I've shown above, there are only a few things left to check. 1. Internal firewall. If you're using a soft-phone on a machine, make sure the firewall is disabled for testing.
No audio for sip calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-for-sip-calls/71048
No audio for sip calls - Asterisk SIP - Asterisk Community. So our asterisk is on an Azure server and we can register the sip phone, but don't hear audio when making calls. We opened ports 15000-19000 on the firewall and changed rtp.conf to start at 15000 and end at 19000. Here i…
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.
How to setup your Asterisk PBX if you are behind a NAT ...
https://my.gradwell.com/s/article/how-to-setup-your-asterisk-pbx-if-you-are-behind-a-nat-firewall
You configure Asterisk choice of RTP ; ports for incoming audio in rtp.conf ; ;nat=no ; Global NAT settings (Affects all peers and users) ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 (;rport) ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport ; (work around more UNIDEN bugs)
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