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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/
NAT = auto_force_rport; if Asterisk can determine that the device is behind NAT, set the force_rport option. The default value, unless the nat option is specified. NAT = auto_comedia; if Asterisk can determine that the device is behind NAT, set the comedia; NAT = force_rport, comedia; option replacing nat = yes in the newer version of Asterisk.
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)
No audio - Fresh install - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-fresh-install/83257
If the endpoint is behind NAT and Asterisk is not configured to know the endpoint is behind NAT, then you will see no audio and can have other places. The options to set to yes would be “rewrite_contact”, “rtp_symmetric”, and “force_rport” if the endpoint itself is behind NAT.
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
Besides NAT problems I've also faced this issues on 3 cases: 1) Missconfigured parameter localnet: 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
Possible newbie issue - No audio on extensions behind NAT ...
https://community.freepbx.org/t/possible-newbie-issue-no-audio-on-extensions-behind-nat-server-tries-to-contact-private-ip/69403
If the extension -> Asterisk path has no audio, something must be wrong unrelated to what the extension is sending in its SDP. If you still have trouble, I recommend setting up a test extension without encryption. If it fails, it’s much easier to debug. If it works, you’ll know the problem is encryption related.
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. Please note that without STUN support, the registrar and proxy server have to be on the same IP.
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
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…
FreePBX behind NAT no audio at all - General Help ...
https://community.freepbx.org/t/freepbx-behind-nat-no-audio-at-all/68819
Hello, Everyone I am new to FreePBX and my box works fine with PSTN. Recently, I applied a sip.us trial account and tried to dial through sip but there is no audio and cdr show as answer. This box is behind NAT and all sip firewall rules applied, There is no sip connection blocked when I made sip call. Please give me some hint to solve this problem. Here is the log > …
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