Iax2 One Way Audio | Audio-Digital.net

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One way audio on an IAX2 trunk | MangoLassi

https://mangolassi.it/topic/9167/one-way-audio-on-an-iax2-trunk

The person on PBX A cannot hear the person on PBX B. PBX B calls to PBX A and both parties have 2 way audio normally. PBX B also has an IAX2 trunk to PBX C and PBX D. All calls work perfectly for those. Here is a test call from PBX A (ext 220) to PBX B (ext 103) where PBX A cannot hear PBX B.

IAX Calls - One Way Audio - narkive

https://asterisk-users.digium.narkive.com/xiZnxo04/iax-calls-one-way-audio

one way audio. This only happens for inbound calls off the PSTN, if. Call comes in from PSTN to site A, gets put into a queue to be. answered by receptionist as site B. Receptionist answers the call, and. then puts the call on hold to perform an attended transfer to an.

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

However, I only have one-way audio (Remote Sites can hear Main Site but Main Site cannot hear Remote Sites) or no-way audio (Remote Sites cannot hear in either direction when they call one another). Remote Sites are able to make calls to and receive calls from the PSTN via the IAX2 trunk to Site 1/ESBC but have one-way audio only (Remote Sites can hear …

Zombie Calls and One way audio - VoIP Forum - Spiceworks

https://community.spiceworks.com/topic/288494-zombie-calls-and-one-way-audio

Usually (almost always) one way audio is because one side or the other is filtering RTP. I would check to see if a firewall is blocking this, you find this also if one side is behind a NAT firewall or router. RTP (part of SIP) uses a dynamic port range between 10.000 and 20.000. You need to make sure these ports are open between your SIP devices.

One way audio: PJSIP over PRI - General Help - FreePBX ...

https://community.freepbx.org/t/one-way-audio-pjsip-over-pri/29851

One way audio: PJSIP over PRI. General Help. ... The problem occurs only on outgoing calls over PRI (no problem when using IAX2 trunk). Asterisk Version - 13.4.0 Freepbx Version - 6.12.65-26 PJSIP version - 2.2 DAHDI version - 2.10.0.1 I know PJSIP is experimental but I have been able to use PJSIP over PRI before.

Looking for a gigabit desk phone that supports iax2 - VoIP ...

https://community.spiceworks.com/topic/461904-looking-for-a-gigabit-desk-phone-that-supports-iax2

You should really have some sort of VPN between the sites, or from the phone to the PBX to guarantee two way audio. Without a VPN or sessions border controller, you will probably experience one way audio. Because the Grandstream GXP 2200 is android based, it supports PPTP, Open VPN, and Neorouter.

[asterisk-users] One way audio half way through call

https://asterisk-users.digium.narkive.com/TFdfHWYJ/one-way-audio-half-way-through-call

I have same problem, but with 1.4 branch, after several minutes, asterisk stops sending packets resulting one way audio, this problem appears especialy when bigger jitter appears (>300ms) on

One-way audio during outbound calls - Asterisk SIP ...

https://community.asterisk.org/t/one-way-audio-during-outbound-calls/82347

Hello All, I have an Asterisk 16 installation which is running behind a TD-LTE modem-router. I have a DID from my ISP which is configured as a SIP trunk (chan_sip). I have set RTP port range to 7000-20000 in Asterisk also have forwarded this port range in my router. Incoming calls from trunk work well. However when dial an outside number, only outgoing …

One-way audio on outgoing calls on Opus - Asterisk Support ...

https://community.asterisk.org/t/one-way-audio-on-outgoing-calls-on-opus/68769

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Exclusive-Senior separatist urges Russia to send 30,000 ...

https://www.thestar.com.my/news/world/2022/02/08/exclusive-senior-separatist-urges-russia-to-send-30000-troops-to-east-ukraine

DONETSK, Ukraine (Reuters) - An influential separatist commander in eastern Ukraine has urged Russia to send 30,000 soldiers to reinforce rebel forces fighting in the breakaway Donetsk region and ...

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