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ONE WAY AUDIO SIP TRUNK - Cisco Community

    https://community.cisco.com/t5/unified-communications/one-way-audio-sip-trunk/td-p/2990120
    That was working okay untill the Firewall at the branch router where the cucm is located was replaced with a cisco router. now there is only one way audio. I have bind commands on the dial peer at the cucm as follows. dial-peer voice 104 voip corlist outgoing VOIP description **Incoming Call from CUCM Trunk** session protocol sipv2

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.

How to punch SIP One Way Audio in the face - UCPros

    https://ucpros.net/sip-one-way-audio/
    we can go ahead and start punching one way audio in the face. Or in more civilized words, troubleshoot one-way audio. Troubleshooting one-way audio and no audio. The process includes three steps: 1. Find where RTP streams are sent to. 2. Verify whether RTP packets are received. 3. Work through possible issues. The logic is simple,

Troubleshooting One-Way Audio - OnSIP Support

    https://support.onsip.com/hc/en-us/articles/204132734-Troubleshooting-One-Way-Audio
    Our network will return the same port for inbound audio as outbound audio, which simplifies the job for the NAT devices involved. Check https://admin.onsip.com. To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail.

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.

Experiencing One Way Audio When Connecting via SIP

    https://knowledgebase.paloaltonetworks.com/KCSArticleDetail?id=kA10g000000PLooCAG
    Experiencing one-way audio when connecting via SIP (Session Initiation Protocol). Environment PAN-OS Cause SIP (Session Initiation Protocol) allows two endpoints to establish media sessions with each other. This is an application layer signaling protocol. The main signaling functions of the protocol are as follows: – Location of an end point.

Avaya Communication Manager Release 6.3 using …

    https://www.cisco.com/c/dam/en/us/solutions/collateral/enterprise/interoperability-portal/avaya-app-note-6-3.pdf
    Configure the trunk group for the SIP trunk to Cisco UCM. Configure the SIP and digital station phone extension. Configure the uniform dialing plan to the Cisco UCM extensions. Configure the route pattern to the Cisco UCM extensions. Cisco Unified Communications Manager: 1. SIP trunk security profile 2. Device setting SIP profile 3.

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.

Issue: One way audio after “call on hold” | Louis UC Blog

    https://ms-uc.herber.co/?p=324
    Further analysis with the trunk provider showed us that this call, while going into hold, sent a sip media flow attribute “sendonly”. This allows only one way audio, so music can be played while preserving bandwith and/or quality. However, when resuming the call, no new invite is sent to cancel the one way audio!

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