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Solved: Problems with SIP Trunk (one way audio) - Cisco ...
https://community.cisco.com/t5/unified-communications/problems-with-sip-trunk-one-way-audio/td-p/2089593
Problems with SIP Trunk (one way audio) Hi everyone ! Our client is testing a new SIP trunk implementation with a different ISP. They have a SIP trunk between a Cisco 2911 and ISP for PSTN accesss, and a H323 Trunk between CUCM ver 7.1.3.30000-1 for proper call routing to that Cisco2911 gateway. Here you have Cisco 2911 configuration:
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
If you are experiencing one-way or no-way / no audio issues, here is what you need to do to fix that easily. ... Note: Cisco Unified Communications Manager only uses 24576-32767 although other devices use the full range ... This example shows a one-way audio, the call flow is SIP phone calls an SCCP phone .
How to punch SIP One Way Audio in the face - UCPros
https://ucpros.net/sip-one-way-audio/
Sometimes you won’t be able to reproduce the one-way audio issue. In that case, you can use the built-in CDR in CUCM to find the direction of the RTP stream. Go to Cisco Unified Serviceability–>Tools–>CDR Analysis and Reporting. Then, CDR–>Search–>By user/phone number/SIP URL find your call and click view on the CDR-CMR dump.
Solved: CUBE One-Way Audio - Cisco Community
https://community.cisco.com/t5/collaboration-applications/cube-one-way-audio/td-p/4076969
CUBE One-Way Audio. Dears, hope you all doing well, We have this environment (call flow): ITSP (SIP Trunk) --> CUBE --> CUCM (5001 is AA) --> CUC (Customer Service 0 is a hunt group on CUCM extension number 1888) We have one way-audio and the call will disconnect after 10-15 seconds when calling to this hunt-group, the weird thing is when I do a "debug ccsip …
CUCM One-way audio and MTP Problems - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/cucm-one-way-audio-and-mtp-problems/td-p/3031464
The one-way audio is not always happening, its intermittent. During normal working calls via the external SIP Provider, all RTP packets are being sent from IP phone to CUCM, which forwards it to the SIP provider. When we have one-way audio issues, the RTP media packets are being sent directly to the SIP provider from the IP Phone.
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
The signaling phase normally traverse intermediary device and/or call processing servers like CUCM, CUCME while the media NORMALLY flows between the two endpoints involved in the call directly, this means that if there must be ip reachability between the two endpoints involved in the call, if there is no reachability at all then you will have no audio or if there is reachability on one …
Cisco Unified Communications Manager Trunks - Cisco
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/session_mgmt/deploy/8_6_1/trunks.html
Inbound calls to Cisco Unified CM from SIP Delayed Offer trunks or H.323 Slow Start trunks that are extended over an outbound SIP Early Offer trunk will use an MTP to create the Offer SDP; however, these calls support audio only in the initial call set up but can be escalated mid-call to support video and SRTP if the call media is renegotiated (for example, …
System Configuration Guide for Cisco Unified ...
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/sysConfig/CUCM_BK_SE5DAF88_00_cucm-system-configuration-guide-1151/CUCM_BK_SE5DAF88_00_cucm-system-configuration-guide-1151_chapter_01110.html
Within Cisco Unified CM Administration, the SIP Trunk Configuration window contains the SIP signaling configurations that Cisco Unified Communications Manager uses to manage SIP calls. You can assign up to 16 different destination addresses for a SIP trunk, using IPv4 or IPv6 addressing, fully qualified domain names, or you can use a single DNS SRV record.
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
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.
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