One Way Audio Problem Asterisk

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One-way Audio over Site-to-Site VPN at Remote Sites ...

https://community.freepbx.org/t/one-way-audio-over-site-to-site-vpn-at-remote-sites/51675#:~:text=One%20way%20audio%20is%20usually%20NAT%20config.%20If,ALL%20the%20local%20subnets%20that%20are%20not%20natted.

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Correcting One-way Audio with a VoIP Call - Asterisk

https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html

One-way audio can occur in either direction, however in-bound audio failure (lack of audio from the outside caller reaching the inside network (LAN) phone) is probably the most common. In many of these cases routers and firewalls could be the …

Asterisk one way audio issue - DIDforSale

https://www.didforsale.com/asterisk-one-way-audio-issue

Are you having an audio issues in your Asterisk? Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues.

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 :

[Help] WebRTC one way audio/video problem on Asterisk 12 ...

https://community.asterisk.org/t/help-webrtc-one-way-audio-video-problem-on-asterisk-12/45335

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Asterisk in Docker - bad one way audio quality - Asterisk ...

https://community.asterisk.org/t/asterisk-in-docker-bad-one-way-audio-quality/79646

The simple stuff works already: I can call linphone on my android mobile with two way audio without any troubles running asterisk in Docker. Check! I’ll update the sip.conf in my original question with the entry of my mobile. So, I think the problem is the WebRTC part. But, I have no clue to what could be the cause. P.s.:

One way audio - FreePBX Community Forums

https://community.freepbx.org/t/one-way-audio/51052

First, try reducing the Local Networks in Asterisk SIP Settings to three entries, as suggested by @dicko. Changing this setting requires you to restart (not just reload) Asterisk. If no luck, examine /etc/asterisk/sip_general_additional.conf and /etc/asterisk/pjsip.transports.conf and confirm that all three entries are present.

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.

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

https://www.voipmechanic.com/voip-one-way-audio.htm

Solve One Way Audio-Step by Step. One way audio where one side or one party can hear the other, but not reverse, is typically indicative of something stopping either the outbound or inbound audio from reaching the receiving party. One way audio can be caused by network issues; NAT (Network Address Translation) issues or firewalls, so finding the cause …

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.

[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.

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