10-28-2013 04:42 AM - edited 03-16-2019 08:06 PM
Hi
We have 3 sites locate in London, Melbourne and New York respectively.
CUCM - 9.1.1.10000-11
VG - 2911 running c2900-universalk9-mz.SPA.151-4.M5.bin
Call flow as bleow:
9971 ----sip-----CUCM -----trunk---- GW ---- PSTN----- ITSP
If Melbourne user call USA it will use TEHO to New York VG for outgoing calls, vice versa.
So does for London and USA.
Recently we suspected that TEHO is not working as expected (I assume it never work properly). User get fast busytone after inter digit timeout straight away. I have done the DNA and it shows call flow is correct and should work.
I have attached the DNA output, isdn q931 debug and CCM trace for failed call.
calling number - 61386725433 (will be translated to other number on VG)
called number - 17193252949
The isdn debug does not tell too much info.
CCM trace looks like it's CUCM send disconnection message to VG to terminate the call.
Any ideas?
10-28-2013 05:30 AM
Hello Fei,
The CUCM disconnects the calls with "reason q.850 cause=47" meaning there is not enough media resource to complete the call. It seems Codec issue. Kindly check the region settings between 'REG-USANYC' and 'REG-AUVICMEL'
79794959.011 |15:53:50.239 |AppInfo |DET-MediaManager-(81230)::preCheckCapabilities, region1=REG-USANYC, region2=REG-AUVICMEL, Pty1 capCount=5 (Cap,ptime)= (4,30) (258,0) (257,30) (259,30) (261,30), Pty2 capCount=6 (Cap,ptime)= (4,20) (2,20) (11,20) (12,20) (6,20) (86,20)
79794959.015 |15:53:50.239 |AppInfo |RegionsServer: applyCodecFilterIfNeeded - no codecs remained after filtering so restored original 0 caps
79794959.016 |15:53:50.239 |AppInfo |DET-MediaManager-(81230)::preCheckCapabilities, caps mismatch! Xcoder Reqd. kbps(8), filtered A[capCount=0 (Cap,ptime)=], B[capCount=2 (Cap,ptime)= (11,20) (12,20)] allowMTP=0 numXcoderRequired=1 xcodingSide=1
79794967.003 |15:53:50.240 |AppInfo |MediaTerminationPointControl(107)::waiting_AllocateMtpResourceReq - (not an error) DeviceCap Mismatch
79794967.026 |15:53:50.241 |AppInfo |MediaTerminationPointControl(107)::SendMTPResourceErrToSender - ERROR AllocateMtpResourceReq failed -- Ci=49707060, errBitset=0x3
79794968.001 |15:53:50.241 |AppInfo |MediaResourceCdpc(50984)::resource_rsvp_AllocateMtpResourceErr Device=TRN-USANYC-01 deviceCapIntersec=256
79794968.003 |15:53:50.241 |AppInfo |MediaResourceCdpc(50984)::adjustDeviceTblXcoder - No enough available resource
79794994.001 |15:53:50.243 |AppInfo |StationD(21332): StationCtiD-CcDisconnReq onBehalf=Media Cause=47 tmpAe.ci=49707056
Please check if the transcoders are registered properly and available in MRGL.
10-28-2013 06:41 AM
Your RTP flow looks like this. Note an MTP device was invoiked because you are using EO.
ip phone------->MTP------->gateway
The inbound dial-peer on the gateway is set to use g711ulaw
From the logs,,,this is the 183 session progress we get back from the gateway to cucm.on the outbound leg of the call
79796878.002 |15:54:28.626 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 172.18.73.254 on port 5060 index 13504 with 1058 bytes:
[52758607,NET]
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 172.18.12.12:5060;branch=z9hG4bK22936757af1347
From: "Jaryd Anning" <61386725433>;tag=16794848~2580bb60-ed88-451a-be78-42a8a117e39f-4970707061386725433>
To: <>;tag=CFC1EA3C-99C>
Date: Thu, 24 Oct 2013 04:49:57 GMT
Call-ID: 5ad70900-2681a803-107008-c0c12ac@172.18.12.12
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <17193252949>;party=called;screen=no;privacy=off17193252949>
Contact: <>>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1271 8975 IN IP4 172.18.73.254
s=SIP Call
c=IN IP4 172.18.73.254
t=0 0
m=audio 26442 RTP/AVP 0 101
c=IN IP4 172.18.73.254
a=rtpmap:0 PCMU/8000------------------------------Here you can see the codec on the dial-peer is g711
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
The region between the MTP device that has been selected for the call and the gateway is set to use g729..as shown from this log..
79796915.011 |15:54:28.628 |AppInfo |DET-MediaManager-(81231)::preCheckCapabilities, region1=REG-USANYC, region2=REG-AUVICMEL
Hence cucm determines that a xcoder is required..
79796915.016 |15:54:28.628 |AppInfo |DET-MediaManager-(81231)::preCheckCapabilities, caps mismatch! Xcoder Reqd. kbps(8)
CUCM tries to allocate a xcoder but coundlt find any tsuitable one.
I suggest you change the inbound dial-peer on your gateway to advertise both g729 and g711 codecs using voice class codec
voice class-codec 1
codec preference 1 g729
codec preference 2 g711alaw
codec preference 3 g711ulaw
You should then remove the codec on the dialpeer and apply the voice class codec. This will remove the need for a xcoder
dial-peer voice xx voip
no codec g711ulaw
voice class-codec 1
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
10-28-2013 08:42 PM
Hi aokanlawon
The voice class codec original configured as below and assigned to all voip dial peer in sh run.txt file attached.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw
codec preference 3 g711ulaw
Regards
Fei
10-29-2013 02:10 AM
Hi sureshsub2
I have checked xcoder configuration, xcoder are registered with CUCM sub and it's included in MRGL assigned to trunk.
The region settings between REG-USANYC' and 'REG-AUVICMEL are using g729.
Regards
Fei
10-29-2013 03:16 AM
OK....
Can we do a quick test. Since you are doing EO on your CUBE there is no need for you to do EO on CUCM. Can you disable EO on your CUCM so we can remove the need for the MTP as this is introducing unneccesary complexity. This way we will have RTP stream flowing directly between the IP phone and the NYC gateway.
Another solution is to configure the region between your MTP (REG-USANYC) and REG-AUVICMEL as G711. This is because your MTP is configured for g711ulaw, but the region between it and the gateway is set to g729..
Let me know which option you had like to try...Once you have made the changes please collect the ff logs from the NYC gateway
1. debug ccsip messages
2. debug voip ccapi inout
3. cucm SDL logs
Can you also include the cucm logs for the test call. Just the
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
10-29-2013 09:06 AM
What is EO?
10-29-2013 09:19 AM
EO stands for early offer. This is a term used in SIP signalling where the SDP (media capabilities) are exchanged with the INVITE. In delayed offer this is exchanged during the 200 OK answer phase
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-07-2013 09:04 PM
Hi aokanlawon
Issue seems resolve by itself from last few days. We have not make any change at all.
Regarding your suggest, we have not able to take it as EO is configured for another purpose - EO over ISDN call.
But something really strange. I noticed that when IOS MTP is UP and register with CUCM, TEHO stop working. But as soon as IOS MTP become unregistered, TEHO start working.
I have attached ccm trace for both scenario. Since I cannot reproduce issue, i was unable to capture debug on VG for failed but able to provide succeed one. 20131108.rar
Final called number as 17193252949.
I have a question, since we do have all 711 and 729 codec has been configured in Xcoder, do we still need MTP invoke for call setup? When MTP invoke?
10-29-2013 04:41 AM
Hi Fei,
As Deji mentioned, is it possible for you to change the codec to G711 between the regions REG-USANYC and REG-AUVICMEL and check the behaviour?
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