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SIP Trunk and MTP Resource Allocation Failure

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This document was generated from CDN thread

Created by: Abdul Rasheed on 01-11-2010 11:22:03 AM
Dear All,
                 Its with great worry that i am posting this thread, we are working on the issue for more than a week and still not able to find a solution and if it is delayed further , we will loose  a big order which will in turn put me in a very bad position  (  . So if anyone can help me, it will be really great.  we have our voice  recorder deployed for an office with some phones in its head office and some of the phones in Branch office.  They are using CUCM 6.1.5  .  Primary and Secondary CUCM is being used.  Following is the scenario of deployment over there. 
 
1. Phones with DN 858 , 852 are deployed in Head office where the CUCM is also placed. 
 
2.     Phones with DN 683, 684 , 685 are deployed in Branch office
 
Following are the recorder configuration and recording results
 
1. Recorder uses SIP trunk with DN 7777 for initiating recording 
 
Following are the scenarios in which call recording is not happening 
 
1.     Calls from Head office from DN 858 or 852 to PSTN is not recorded
 
 2.    calls from  Head office from DN 858 to DN 852 is not recorded  
 
3.     Calls from Head office from DN858/852 to DN 683/84/85  in branch office is not recorded. 
 
4.    Call from Branch office from DN 683/84/85 to DN858/852 is not recorded
 
Calls which are recorded includes the following
 
1. Out going CAlls from Brnach office DN 683/84/85 to any PSTN
 
2..  Incoming calls to DN 683/84/85 from any PSTN number
 
3.    Incoming calls to DN 858/852(Head office) from any PSTN number
 
 
As a summary , All the incoming calls to head office through PSTN is recorded, All the incoming/outgoing calls from Branch office are recording except calls between headoffice and branch office. 
 
When i checked the wire shark traces  i could find that the Recorder is receiving a BYE from CUCM with Reason Q.850 cause 47 Resources not available.  When i checked RTMT logs from Cisco following logs were recieved.  From TAC i got the information that the MTP resources should be added. But i am not sure how to add it and based on what critirias.  It will be great if somebody can advise me on this. 
 
 

024368560| 2010/10/18 12:45:43.778| 002| SdlSig |
MrmAllocateXcoderResourceReq | waiting | MediaResourceManager(2,100,139,1)|
MediaManager(2,100,141,67005) | (2,100,185,1).2089-(*:192.168.200.3) | [R:NP -
HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0 DBP: 0] CI=36034362 MRGL= Kpbs=0
RegionA=millenium CapA=1 RegionB=millenium CapB=5 SuppressFlag=0 Type=1
DeviceCapablity=0
024368561| 2010/10/18 12:45:43.778| 002|
AlarmErr | | | | | | AlarmClass: CallManager, AlarmName:
MtpNoMoreResourcesAvailable, AlarmSeverity: Error AlarmMessage: ,
AlarmDescription: No more MTP resources available., AlarmParameters: AppID:Cisco
CallManager, ClusterID:StandAloneCluster, NodeID:Subscriber,
024368562| 2010/10/18 12:45:43.778| 002| SdlSig |
MrmAllocateMtpResourceErr | waitResourcesAllocated |
MediaManager(2,100,141,67005) | MediaResourceManager(2,100,139,1)|
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP:
0 DBP: 0] CI=36034362
024368572| 2010/10/18 12:45:43.779| 002| SdlSig | AuConnectErrorInd | connecting |
MatrixControl(2,100,143,61197) | ConnectionManager(2,100,171,1) |
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP:
0 DBP: 0]CI1=36034357 CI2=36034358
clearType=52693512
024368575| 2010/10/18 12:45:43.779| 002| SdlSig | ConnErr |
tcc_await7 | Cdcc(2,100,175,61201) | MatrixControl(2,100,143,61197) |
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 1, LP: 0, VLP: 0, LZP:
0 DBP: 0]


024368577| 2010/10/18 12:45:43.779| 002| SdlSig | CcDisconnReq
| restart0 | BuiltInBridgeControl(2,100,179,382)| Cdcc(2,100,175,61201) |
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 2, LP: 0, VLP: 0, LZP:
0 DBP: 0]CI=36034357 CI.branch=0 clearType=0
c.l=1 c.cid=8 c.cs=0 c.lc=0 c.r=0
cv=47FDataType=0opId=0ssType=0invokeId=0resultExp=0 OnBehalf=Media rfr=0
rdDestPart= rdDestPatt= rdDestCdpn=pi=0si1 unModRDDestCdpn=pi=0si1
redDestName=locale: 1 Name: UnicodeName: pi: 0


024368578| 2010/10/18 12:45:43.779| 002| SdlSig | CcDisconnReq
| restart0 | SIPD(2,100,86,5) | Cdcc(2,100,175,61201) |
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 3, LP: 0, VLP: 0, LZP:
0 DBP: 0]CI=36034358 CI.branch=0 clearType=0
c.l=1 c.cid=8 c.cs=0 c.lc=0 c.r=0
cv=47FDataType=0opId=0ssType=0invokeId=0resultExp=0 OnBehalf=Media rfr=0
rdDestPart= rdDestPatt= rdDestCdpn=pi=0si1 unModRDDestCdpn=pi=0si1
redDestName=locale: 1 Name: UnicodeName: pi: 0
024368584| 2010/10/18 12:45:43.779| 002| SdlSig | CcDisconnReq
| connect_request8 | SIPCdpc(2,100,87,234) | SIPD(2,100,86,5) |
(2,100,185,1).2089-(*:192.168.200.3) | [R:NP - HP: 0, NP: 2, LP: 0, VLP: 0, LZP:
0 DBP: 0]CI=36034358 CI.branch=0 clearType=0
c.l=1 c.cid=8 c.cs=0 c.lc=0 c.r=0
cv=47FDataType=0opId=0ssType=0invokeId=0resultExp=0 OnBehalf=Media rfr=0
rdDestPart= rdDestPatt= rdDestCdpn=pi=0si1 unModRDDestCdpn=pi=0si1
redDestName=locale: 1 Name: UnicodeName: pi: 0
024368589| 2010/10/18 12:45:43.779| 002| SdlSig | SIPDisconnReq
| wait | SIPHandler(2,100,84,1) | SIPCdpc(2,100,87,234) |
(2,100,185,1).2089-(*:192.168.200.3) | [T:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP:
0 DBP: 0] CcbId= 263 --TransType=2 PeerAddr = 192.168.200.3:5060 ccCause= 47 sip_disc_cause= 0
10/18/2010
12:45:43.781 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to
192.168.200.3:[5060]:
BYE sip:7777@192.168.200.3:5060
SIP/2.0
Reason: Q.850;cause=47
Date:
Mon, 18 Oct 2010 08:45:43 GMT
From: "Mustafa Al-Ani" <sip:858@192.168.200.2;x-nearend;x-refci=36034353;x-nearenddevice=SEP001BD47DC9F3>;tag=89c7999e-90d4-4307-985c-3b62e8ae3562-36034358
Content-Length:
0
User-Agent: Cisco-CUCM6.1
To: <sip:7777@192.168.200.3>;tag=ccdbb47a-2106-4cb4-9cb4-f3b0a7c6d6d7-20026578
Call-ID:
165c1b00-cbc10937-ea-2c8a8c0@192.168.200.2
Via:
SIP/2.0/UDP 192.168.200.2:5060;branch=z9hG4bK29451de3f91
CSeq: 102
BYE
Max-Forwards: 70
 
 
 
 
Thanks and Regards
Rasheed

Subject: RE: SIP Trunk and MTP Resource Allocation Failure
Replied by: Giggesh Thekkekeloth on 03-11-2010 07:38:16 AM
Hi,

I believe this issue is rellated to the SR :http://wwwin-tools.cisco.com/casekwery/getServiceRequest.do?id=615699607 which was handled by me.

where, we have diagonized that,
when the media is setup for the recording call,e.g. the call from BIB to the recorder, the media is asking for the MTP but MTP is not available. Then the recording call was torn down and that's why BYE was sent to the recorder over the sip trunk.

You said, you would open a TAC case to assist you on setting up the MTP, do you need any help on opening the TAC case?


My search has got to this page which states about 'Media Resources: Media Termination Point Parameters'
http://www.cisco.com/en/US/solutions/collateral/ns340/ns517/ns431/SIP_trunk__SRND.pdf

Subject: RE: SIP Trunk and MTP Resource Allocation Failure
Replied by: Abdul Rasheed on 03-11-2010 12:41:37 PM
Dear Giggesh,
                         Thanks a lot. i have already opened a new TAC on this SR615797951  and he accessed the system a couple of times and even tried adding the MTP resources also, but still the problem is persisting. we are now taking some ethereal traces and CUCM logs of the working and non working scenarios.   But i am under much pressure from Customer to finish the project soon. So i was just checking if somebody who already faced this issue can help me on this for finding a quick solution if possible.  
 
This customer deployment  has 2 locations  1 head office and 1 branch office.  All the incoming/outgoing calls through PSTN from branch office is getting recorded.  All the Incoming PSTN calls to head office is recording.  But the calls between 2 offices,  all out going calls from Head office,  all internal calls between extensions in head office etc are not recorded.   So we are confused,  to whcih region we should add the MTP resources.  we have enabled a recording  tone  through service parameters. I am not sure if that is taking an additional MTP resource. Tomorrow i am going to disable the tone during recording and try again. 
 
Thanks and Regards
Rasheed


 
 
Hi,

I believe this issue is rellated to the SR :http://wwwin-tools.cisco.com/casekwery/getServiceRequest.do?id=615699607 which was handled by me.

where, we have diagonized that,
when the media is setup for the recording call,e.g. the call from BIB to the recorder, the media is asking for the MTP but MTP is not available. Then the recording call was torn down and that's why BYE was sent to the recorder over the sip trunk.

You said, you would open a TAC case to assist you on setting up the MTP, do you need any help on opening the TAC case?


My search has got to this page which states about 'Media Resources: Media Termination Point Parameters'
http://www.cisco.com/en/US/solutions/collateral/ns340/ns517/ns431/SIP_trunk__SRND.pdf


Subject: RE: SIP Trunk and MTP Resource Allocation Failure
Replied by: Carlos Pinilla Arbex on 24-11-2010 09:27:36 AM
Hello,
 
We are having a similar problem. SIP Trunk and MTP have been checked and they work fine when an external call goes directly to a phone; MTP resources are allocated correctly. If we try to transfer the call from the phone to CUAE, it fails and the call is disconnected.
 
When the call goes to CUAE instead a phone, no MTP resources are allocated but the script seems to work fine; MTP is requiered but not allocated. It seems like CUAE is working like MTP. If we try to transfer the call from CUAE to a phone, it fails and the call is disconnected.
 
Any idea?
 
Thanks & Regards,
Carlos

Subject: RE: SIP Trunk and MTP Resource Allocation Failure
Replied by: Abdul Rasheed on 25-11-2010 07:54:34 AM
Dear Carlos,
                       Do you have G.722 Codec Advertised in the Enterprise parameters?  If yes , are you negotiating G.722 in your SDP for Codec Capabilities.?    Our issue was solved by  disabling the option "Advertise G.722 codec " from CUCM enterprise parameters as because we were not negotiating G.722 in our codec capabilities.    Can you please check this possibility and please let me know if this solve the issue. I am also anxious to know the results  . 
 
Thanks  and Regards
Rasheed
 
 
 
Hello,
 
We are having a similar problem. SIP Trunk and MTP have been checked and they work fine when an external call goes directly to a phone; MTP resources are allocated correctly. If we try to transfer the call from the phone to CUAE, it fails and the call is disconnected.
 
When the call goes to CUAE instead a phone, no MTP resources are allocated but the script seems to work fine; MTP is requiered but not allocated. It seems like CUAE is working like MTP. If we try to transfer the call from CUAE to a phone, it fails and the call is disconnected.
 
Any idea?
 
Thanks & Regards,
Carlos

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