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Correcting One-way Audio with a VoIP Call - Asterisk

    https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
    SIP transformations are known to corrupt some of the SIP headers resulting in issues with the transfer of the voice traffic correctly. Follow these basic steps to isolate the cause of one-way audio: Connect the ATA/IAD or other SIP device …

One way audio problem in trunk - Asterisk SIP - Asterisk ...

    https://community.asterisk.org/t/one-way-audio-problem-in-trunk/72567
    Hello. I got some problem for Asterisk SIP trunk. It’s Asterisk 11.17.1. SIP trunk from an operator. Outgoing call : Everything is ok Incoming call : Call signaling is ok but audio RTP is one way. Peer A,B and C can send audio packet to Peer X but it isn’t able to receive audio packet from Peer X. Network :

How to solve VoIP one way audio- step by step.

    https://www.voipmechanic.com/voip-one-way-audio.htm
    If one way audio still exists check to see if you have a public or private (192.168.1.xxx) IP address. Public IP- Call your VoIP provider. If you are getting one-way audio with a public IP address, there is an issue with the way the VoIP provider is handling the call.

sip - Asterisk one way audio - Stack Overflow

    https://stackoverflow.com/questions/55643399/asterisk-one-way-audio
    Trying to call from a sip client to a normal phone or exetension. This results always in a one way audio connenction. I use the odbc database, and can't really find the problem. Can anybody help me in the right direction. There seems to be no errors at all. Have tried several things, and searched on the net, coudn't find the correct solution.

One way audio for sip etensions - General Help - FreePBX ...

    https://community.freepbx.org/t/one-way-audio-for-sip-etensions/7365
    I have an Asterisk@Home installation in my organization. I have had this installation for more than 3 years but never noticed this problem. There are 2 problems, but both seems to be related. My SIP extensions have one way audio SIP extensions have one way audio; others can hear them but not vice-versa. No, this is not a problem with one desktop. This is …

[SOLVED] One way audio - VoIP Forum - Spiceworks

    https://community.spiceworks.com/topic/388912-one-way-audio
    Typically one way audio has to do with filtering (e.g. QoS) of the audio stream. You'll see the port negotiation in the original SIP packets during the handshake. Then just check for incoming packets on that port. If you don't see any, check upstream for the blocking router.

One way audio - FreePBX - FreePBX Community Forums

    https://community.freepbx.org/t/one-way-audio/51052
    When I move a phone into one of the VLANs, it registers fine with FreePBX, but audio will only flow one way (from the external phone to the VoIP phone). I have captured packets on the firewall, and it appears that the audio from the VoIP phone is being sent to the external public IP instead of the internal private IP.

One way audio – Yeastar Support

    https://support.yeastar.com/hc/en-us/community/posts/360001248568-One-way-audio
    One way audio. Kris Vandewiele April 17, 2018 09:44. ... SIP ALG tries to do some packet inspection and adds routing to clearly identified SIP packets including RTP packets; if you're using it your firmware on your router should be all the way updated to whatever they last released. I personally make it a troubleshooting step to disable SIP ALG ...

How to troubleshoot one-way and no-way audio on VoIP …

    http://info.teledynamics.com/blog/how-to-troubleshoot-one-way-and-no-way-audio-on-voip-calls
    One security component of a firewall specifically designed for voice is the SIP Application Layer Gateway (ALG). SIP ALG is a firewall setting that can either be enabled or disabled -- generally, the audio issues occur when it's enabled. (3) Incompatible Codecs

Pfsense and Asterisk….one-way audio problem with SIP ...

    https://forum.netgate.com/topic/465/pfsense-and-asterisk-one-way-audio-problem-with-sip
    This shows up on voicemail as well...Asterisk hangs up immediately as it cannot hear any audio. The inbound caller can hear me speaking. My thoughts are that this is the classic 'one-way audio over SIP' problem, which can be found on certain firewall types/configurations ie symmetric firewalls. The static port feature may be the way around this.

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