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Feature Code Call Transfers - Asterisk Project - Asterisk ...
https://wiki.asterisk.org/wiki/display/AST/Feature+Code+Call+Transfers#:~:text=ALICE%20decides%20to%20transfer%20BOB%20to%20extension%20103%2C,channel%20for%20BOB%20that%20is%20dialing%20extension%20103.
Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
Connect the ATA/IAD or other SIP device directly to the first device on the LAN such as the modem. Check for normal operation and good two-way audio. If you experience one-way audio or do not receive dial tone, then the issue will most …
Asterisk one way audio issue | DIDforSale
https://www.didforsale.com/asterisk-one-way-audio-issue
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 codec on your switch. For example if you are using G729 then remove ULAW ...
PJSIP One Way Audio - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-one-way-audio/84926
Hi, After spending an hour on the phone with my providers, it seems that I have a problem with my settings for my endpoint or my transport. Where ? we did not find… So we have a concern for one way audio. If I call an external number from my asterisk, the person hears me, but I cannot hear the person. Config : [transport-udp] type=transport protocol=udp bind=0.0.0.0 …
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 - not all services - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/one-way-audio-not-all-services/83048
Hello, I have no issues with asterisk and now I have one more difficult. what observed: One way audio is for outgoing and incoming calls. External calls. missing stream is always outgoing from my internal phones I use trunk I checked IVR calling by GSM and I was able to hear it I checked conference bridge - one caller internal one external and then everthing was …
Asterisk - One way audio with PJSIP over PRI - Stack …
https://stackoverflow.com/questions/31031273/asterisk-one-way-audio-with-pjsip-over-pri
Update: I registered a soft phone (Sipdroid) to the PJSIP extension and tried dialing out through the PRI trunk and everything worked. So I thought maybe the problem is the phone itself (Yealink T48g), took a new phone out of the box (Yealink T28p) with the same version and settings as I have running PJSIP for my other client (PBX is also exactly the same build) and …
One way audio after 3-6 minutes of established call - FreePBX
https://community.freepbx.org/t/one-way-audio-after-3-6-minutes-of-established-call/18351
I am experiencing one way audio after 3-6 minutes of established call. Here is the network settings The NAT setting of Asterisk is disabled. The network of SIP client connect to the network of PBX thru PPTP VPN. The subset of the client network is 192.168.6.x and the PBX side is 192.168.10.x The client side is using Grandstream 1450 and the server side is using Trixbox …
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