I have the follow problem:
PSTN----Cisco CUBE Sip Trunk--- CUCM----UCCX
When calls form PSTN there no audio from UCCX IVR. The calls remain for a moment and then are disconnected.
Please see the attach. with deb ccapi - deb ccsip message - deb ccsip calls and media.
Calling number is: 226861431
Called number is: 2450640
UCCX Trigger is : 1380
IP CUCM: 10.56.240.57
IP CUBE: 172.16.22.81
Seems either MTP is already enabled or because of some other reason, CUCM is sending CUBE IP 172.16.22.81 in 'c' line under 200 OK SDP answer. If MTP wouldn't have been enabled, CUCM should have sent UCCX IP address in 'c' line under 200 OK SDP answer.
Along with what Ayodeji said, you should also share how your SIP Trunk and CTI Port configuration looks in CUCM.
You are right, but this may not be because mtp is enabled on the sip trunk. It's most likely due to dtmf mismatch since uccx ports do not support rfc2833.
Best way to determine what's going on is to look at cucm logs..
Can you please send cucm logs for the period of your test
We need cucm logs.. Please us the link below to collect the logs
Inlcude the calling and called number and time of call..
Please open up the logs and ensure the trace for the call is present. If you have multiple nodes in your cluster collect the logs from all..
Okay here is my observation from your logs..
1. +++ CUCM invokes MTP due to DTMF mismatch between CUBE and CTI Port_1383 +++
01593825.012 |11:00:33.280 |AppInfo |DET-MediaManager-(59743404)::isMTPNeededForMismatchOrConfig, MTPNeededDueToDTMFCapMismatch(2833/OOB) mtpinsertionReason=1 dtmfMTPSide=1
2. +++ CUCM allocates MTP for dtmf mismatch +++
MediaTerminationPointControl(833)::getResourcesAllocated -- DeviceName=MTP24E9B3C43B40 Ci=60677010 ResourceAllocated=1
3. +++ CUCM exchanges OLC and OLC ACK between CTI_Port and MTP +++
MediaTerminationPointControl(833)::star_StationOutputOpenReceiveChannel - TCPPid = [188.8.131.52447617] myIP: 0x511610ac (172.16.22.81) ConferenceID: 62714836, MediaPartyId: 65846813, msecPacketSize: 20 compressionType: 4
4. +++ finally media transmission starts between MTP and CTI_port +++
MediaTerminationPointControl(833)::star_StationOutputStartMediaTransmission - TCPPid = [184.108.40.206447617] myIP: 0x511610ac (172.16.22.81)
01593981.002 |11:00:33.286 |AppInfo |MediaTerminationPointControl(833)::star_StationOutputStartMediaTransmission - ConferenceID: 62714836, MediaPartyId: 65846813, RemoteIpAddr: 0x3cf0380a (10.56.240.60) RemoteRtpPortNumber: 28552 msecPacketSize: 20 compressionType: 4
5. +++ Finally CUCM sends 200 OK with ip address of MTP for media +++
01594016.001 |11:00:33.299 |AppInfo |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.22.81::
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.22.81:5060;branch=z9hG4bK141D887
o=CiscoSystemsCCM-SIP 269784370 1 IN IP4 10.56.240.57
c=IN IP4 172.16.22.81
m=audio 32158 RTP/AVP 0 101
So based on this this is what your media flow should look like..
CTI_Port 1383---RTP-----MTP (CUBE)-------RTP------MTP(CUBE)
The first question to ask since you are not getting media is this..
Can the IP subnet of CTI_Port (10.56.240.60) reach the cube IP subnet ??
Is there any firewall between these two subnets?
The resources in the router are the problem (MTP and transcoder).
When we hang up the call, the resources are held in the router and don't close. We think that can be a ios Bug.
In show sccp all, there are many errors for MTP and Transcoder resources.
If we restart the router, the resources are closed and the calls work in the right way (audio and dtmf).
If you look at the logs closely, you will notice that the 200 OK sent to the ITSP doesn't include any dtmf attributes.
Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKmur0uk00d051pr8me6p1.1 From: <sip:firstname.lastname@example.org:5060;user=phone>;tag=eh3co3hk-CC-27 To: <sip:email@example.com:5060;user=phone>;tag=4E0BD450-90B Date: Fri, 02 Oct 2015 18:49:37 GMT Call-ID: uckdugodlhleb3fmbmofpdu3bdtptbmh@SoftX3000 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:firstname.lastname@example.org>;party=called;screen=yes;privacy=off Contact: <sip:email@example.com:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-15.2.4.M5 Supported: timer Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 191 v=0 o=CiscoSystemsSIP-GW-UserAgent 5118 4846 IN IP4 172.16.22.81 s=SIP Call c=IN IP4 172.16.22.81 t=0 0 m=audio 30780 RTP/AVP 0 c=IN IP4 172.16.22.81 a=rtpmap:0 PCMU/8000 a=ptime:20