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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#:~:text=You%20configure%20Asterisk%20choice%20of%20RTP%20%3B%20ports,send%20rport%20%3B%20%28work%20around%20more%20UNIDEN%20bugs%29
One way audio, no NAT - Asterisk Support - Asterisk Community
https://community.asterisk.org/t/one-way-audio-no-nat/81009
One way audio, no NAT. Asterisk. Asterisk Support. Furious_George September 24, 2019, 12:49am #1. ... X-Lite is not sending audio to Asterisk (or it’s getting blocked), Asterisk is forwarding the media received from the gateway to X-Lite at the IP address and port it gave. Your problem is likely external to Asterisk, be it a firewall or ...
SIP Trunk, No NAT, One Way Audio - Asterisk Support ...
https://community.asterisk.org/t/sip-trunk-no-nat-one-way-audio/33583
Hi, Asterisk 1.6.2.11. SIP trunk from an operator. Outgoing call : signal is OK, audio is only one way. I can hear the distant person, but she can’t hear me. Networking : [IP Phone] ------- [Asterisk] ------- [VoIP Provider's Router] ------- [VoIP Provider's PABX] 192.168.4.107 eth0 : 192.168.4.1 eth1 : 37.1.1.1 37.1.1.2 212.39.140.250 pabx*CLI> sip show peers altitude-output …
Asterisk one way audio issue | DIDforSale
https://www.didforsale.com/asterisk-one-way-audio-issue
nat=yes canreinvite=no disallow=all allow=ulaw&g729 context=from-trunk dtmfmode=auto. The Second possible reason for causing one way audio could possibly be Codec, This often happens when a call comes in with ULAW and the system tries to accept with other codecs which can cause superfluous codec negotiation. To avoid this we need to remove the …
Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
If you experience one-way audio or do not receive dial tone, then the issue will most likely be SIP unfriendly NAT, or a firewall present that is not allowing the correct VoIP packets to cross. Try setting up port forwarding or IP tunneling. This will send the needed packets directly to the IAD/ATA device.
Intermittent One-Way Audio -- No NAT - General Help ...
https://community.freepbx.org/t/intermittent-one-way-audio-no-nat/51375
Hi, Running FreePBX 14.0.3.6 with Asterisk 13.19.1 and later upgraded to Asterisk 13.21.1 I am having intermittent one-way audio problems. There is no NAT involved and I run VPN’s from each site back to the datacenter with the PBX. I have done packet captures at the FreePBX machine and see the SIP/SDP ports the RTP audio should be sent to however the PBX …
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).
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.
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 Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
SIP One Way Audio Troubleshooting. Once these ports have been forwarded to the IP of your Asterisk server, give your router a reboot. Re-Login to your router just to reconfirm that the posts are in fact pointing to the correct IP of the Asterisk Server. The next step is to ensure that you configure your NAT settings on the Asterisk server correctly.
How to solve VoIP one way audio- step by step.
https://www.voipmechanic.com/voip-one-way-audio.htm
One way audio can be caused by network issues; NAT (Network Address Translation) issues or firewalls, so finding the cause will require simplification of the connection by elimination of some equipment, then testing. Once the problem device is found its pretty much initiating changes that resolve the issue.
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