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Asterisk one way audio issue | DIDforSale

    https://www.didforsale.com/asterisk-one-way-audio-issue
    asterisk sip trunk provider. Some of our key SIP Trunking features:- ... 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 unwanted ...

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 :

SIP Trunk, No NAT, One Way Audio - Asterisk Support ...

    https://community.asterisk.org/t/sip-trunk-no-nat-one-way-audio/33583
    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.

One way audio help - Asterisk SIP - Asterisk Community

    https://community.asterisk.org/t/one-way-audio-help/88526
    I have a FreePBX 15.0.16.38 and asterisk 16.15.0, I have pjsip extensions and a sip trunk, randomly when I call, I have only one way audio, they hear me but I don’t hear anything. I thought it was a nat problem and I have configured the trunk as follows. type = peer secret = ECOxxxxxxxxx qualify = yes progressinbound = yes nat = force_rport, comedy insecure = port, …

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 in outbound calls to trunk - Providers ...

    https://community.freepbx.org/t/one-way-audio-in-outbound-calls-to-trunk/64083
    Hello All, I am running FreePBX 15 and 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 SIP settings. Also have forwarded this port range in my router. Incoming calls from trunk work well. However when dial an …

[SOLVED] Remote extension, one-way audio on sip client ...

    https://community.freepbx.org/t/solved-remote-extension-one-way-audio-on-sip-client/65376
    If you still have trouble, at the Asterisk command prompt, type sip set debug peer 1234 (replace 1234 with the extension number of the Linphone user, with Linphone configured without STUN) then make a test call that fails with one-way audio. Paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here.

[SOLVED] One way audio - VoIP Forum - Spiceworks

    https://community.spiceworks.com/topic/388912-one-way-audio
    Can you get a packet capture ( Wireshark) in front of the SIP trunk? 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.

Solved Yet another one way audio post, outbound SIP …

    https://www.3cx.com/community/threads/yet-another-one-way-audio-post-outbound-sip-trunk-calls-called-party-can-not-hear.62013/
    External calls to our SIP trunk have one-way audio, the called party can not her me but I can hear them. Here's some information... On premise 3CX version 15.5.15502.6 One SIP trunk with SIP.US PBX Delivers Audio is checked on the trunk Firewall tests all pass Firewall is Fortigate FG100D v6.0.3 build0200 (GA) SIP ALG is disabled per this document:

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 …

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