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SIP trunk - CUBE - one way audio - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/sip-trunk-cube-one-way-audio/td-p/2783599
SIP trunk - CUBE - one way audio Hi guys, I am having a problem deploying a new SIP trunk to a customer. The scenery is the following: PSTN———>GW (CUBE)————>CUCM——----->IP PHONE. PSTN = Internet link with SIP trunk. GW (CUBE) = Cisco 2901 router. CUCM = CallManager 8.6.
SIP trunk with one way audio - Cisco Community
https://community.cisco.com/t5/other-collaboration-subjects/sip-trunk-with-one-way-audio/td-p/315058
SIP trunk with one way audio. Hello all, I have a problem with a SIP trunk in CallManager 4.0.2a. I have set up a SIP trunk (UDP and alaw codec). The SIP device is a Vegastream 400. When I call to a SIP phone from a CallManager phone, I can only hear voice from SIP side to CallManager side. There is no voice from CallManager phone to SIP side.
How to punch SIP One Way Audio in the face - UCPros
https://ucpros.net/sip-one-way-audio/
To verify that your SIP Trunk has a SIP-Profile with SIP Rel1xx set to ‘Send PRACK if 1xx Contains SDP’. There were also reports the command. voice rtp send-recv. helped in some cases of silence and one way audio to the PSTN, when there was no ISDN Connect message, so don’t hesitate to try it.
How to troubleshoot one-way / no audio issues - Cisco ...
https://community.cisco.com/t5/collaboration-voice-and-video/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442
How to check really quick if the phones are sending / receiving RTP (audio). * Open the web page for 2 test phones, then click the 'stream 1' link located at the left handed side of the page, and check if the IP address and port match the information on both sides, keep pressing the 'stream 1' link and you will notice that the Tx and Rx stats ...
Shoretel to Cisco SIP trunk one way audio - ShoreTel Forums
https://www.shoretelforums.com/forum/shoretel-tech/installers/1743-shoretel-to-cisco-sip-trunk-one-way-audio
Shoretel to Cisco SIP trunk one way audio 12-13-2008, 09:21 AM. Where are having a buggy issue with our SIP trunks between Cisco Callmanager and Shoretel. We can make/receive phone calls perfectly and have no audio issues. The problems occurs when a user on a shoretel phone puts the Cisco user on hold, and then tries to pick the call back up ...
How to resolve one-way or no-way audio on VoIP calls
http://info.teledynamics.com/blog/how-to-resolve-one-way-or-no-way-audio-on-voip-calls
One of the most common challenges involves a technology called Network Address Translation (NAT), which for data networks has been a godsend, but if not configured carefully, could cause problems for voice applications. In particular, NAT is a common cause of one-way and no-way audio on VoIP calls.
SIP trunk intermittent one way audio
https://social.technet.microsoft.com/Forums/en-US/f3d358c9-e4d6-490b-8f8b-9fa28c30a489/sip-trunk-intermittent-one-way-audio
I have looked at the flow charts for the calls and do no see any odd BYE request or re-invites during the call that would correspond with the audio disconnecting. I also want to mention that we have an ATA device with direct SIP trunk connection that does not go through our lync servers at all and still experience the one way audio.
No Audio or One Way Audio after Transfer or Forwarding - Cisco
https://learningnetwork.cisco.com/s/question/0D53i00000qeorH/no-audio-or-one-way-audio-after-transfer-or-forwarding
I am new to Cisco Collaboration and settin up IP Phones. Incoming call from PSTN(Yeaster200). once i click the transfer button on the IP phone for the incoming call. Once call is transferred and picked on phone B, it would have the one-way audio or no audio on both sides for the caller and the transfer target party.
Avaya Communication Manager Release 6.3 using SIP trunk …
https://www.cisco.com/c/dam/en/us/solutions/collateral/enterprise/interoperability-portal/avaya-app-note-6-3.pdf
Cisco UCM – SIP trunk to Cisco IM&Presence Trunk Configuration ..... 131 Acronyms ... One way audio between Avaya video phones when Cisco UCM transfers the video Call. Cisco UCM sends re-INVITE after the redirection and updates SDP but Avaya is not updating the SDP resulting
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