Call Disconnect from a remote destination mobile number phone to an internal extension

[alemunmo81]

Hello All

I am having an issue of calling disconnect on the CUCM when some mobile phones call to the auto-attendant (unity connection), and also these mobile phones calls are rerouted to any internal extension. One more thing, these mobile phone calls are remote destination configured on the CUCM.

The flow of the the call is the following:

Remote Destination Mobile Phone Number --> SIP Trunk PSTN --> Gateway 2921 --> AA (Unity Connection) --> Internal Extension Number.

On the other hand, when i have done the call not using the SIP Trunk PSTN otherwise a PRI Interface, it works well.

Any idea?. I would appreciate the help.

Thanks,

Alexis

[Ayodeji Okanlawon]

We need to look at some logs. Can you send "debug ccsip messages". Please send calling and called number

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[alemunmo81]

Hello Aok

I attach the debug requested for your analysis.

I would appreciate your help with this.

Thanks,

Alexis

[Harmit Singh]

Hi Alexis,

Having the CUCM SDI and SDL traces would have really helped along with the debugs. Having said that, I have gone through them and I see a BYE coming from 192.168.11.162:

6685792: Nov 24 10:26:17.084: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:+12027024513@192.168.11.3:5060 SIP/2.0

Reason: Q.850;cause=47

Date: Sat, 24 Nov 2012 15:25:55 GMT

From: <700>;tag=80fddeeb-94b3-4b1c-b9fe-8524fe2a75d8-34683081

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <>;tag=FE2E8BF8-A8

Call-ID: 132BDF97-358211E2-825FD74D-21E74B95@192.168.11.3

Via: SIP/2.0/UDP 192.168.11.162:5060;branch=z9hG4bK445321793dc0c5

CSeq: 103 BYE

Max-Forwards: 70

Cause 47 signifies a resource issue. What is 192.168.11.162? I'm assuming its the CUCM node by the looks of it. I can see that Gateway and CUCM  as well as Gateway and service provider negotiate G729 codec.

Action Plan:

-----------------

We need details CUCM SDI and SDL traces with SIP call processing and SIP Stack options enabled. Please disable SIP Stack once the test call is captured. Also, please provide IP address and hostnames of all UCM nodes in the cluster, gateway IP, provider IP, Unity Connection IP, timestamp of test call, calling, called party numbers, etc to analyze the traces.

I would also check what is the region setting between internal extension and SIP trunk to gateway, Voicemail port and internal extension, etc. If any call leg is G711, then a trancoder would be needed, which can cause the call to fail with cause=47 if it is not available.

I've attached the entire exchange of SIP messages for your reference.

HTH.

Regards,

Harmit.

[alemunmo81]

Hello

I have a doubt, how can i get these logs using CUCM SDI and SDL traces with SIP call processing and SIP Stack options enabled??...should i use the real time monitoring tool?

I appreciate the help.

Thanks,

Alexis

[Ayodeji Okanlawon]

Hi,

Yes I agree with Harmit..

The bye is coming from cucm...

6685792: Nov 24 10:26:17.084: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:+12027024513@192.168.11.3:5060 SIP/2.0

Reason: Q.850;cause=47

Date: Sat, 24 Nov 2012 15:25:55 GMT

From: <700>;tag=80fddeeb-94b3-4b1c-b9fe-8524fe2a75d8-34683081

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <>;tag=FE2E8BF8-A8

Call-ID:

132BDF97-358211E2-825FD74D-21E74B95@192.168.11.3

Via: SIP/2.0/UDP 192.168.11.162:5060;branch=z9hG4bK445321793dc0c5

CSeq: 103 BYE

Max-Forwards: 70

:
We will need CUCM logs to investigating why CUCM is sending a bye.

Cause code 47 most times usually refer to Xcoder, MTP or firewall related issues...

What is the region configured between the Gateway and the IP phones.

The dial-peer configured to send calls to CUCM from the gateway is set to use G729 as we see from the m-line of the trace

Sent:

INVITE sip:700@192.168.11.162:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.11.3:5060;branch=z9hG4bK22AF14228B

From: <>;tag=FE2E8BF8-A8

To: <700>

Date: Sat, 24 Nov 2012 15:25:55 GMT

Call-ID:

132BDF97-358211E2-825FD74D-21E74B95@192.168.11.3

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 270

v=0

o=CiscoSystemsSIP-GW-UserAgent 1072 4723 IN IP4 192.168.11.3

s=SIP Call

c=IN IP4 192.168.11.3

t=0 0

m=audio 18396 RTP/AVP 18 101

c=IN IP4 192.168.11.3

a=rtpmap:18 G729/8000

In this case if the region between the gateway and the phone is set to use g711, then you will need a xcoder provisioned on the MRGL of the gateway. The region between the xcoder and the gateway must also be set to use g711.

I wrote a document on the interaction between regions and codecs on a sip gateway..have a read here...

https://supportforums.cisco.com/docs/DOC-26098

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[alemunmo81]

Thanks all for yours useful answers, i am gonna check carefully each suggestion delivered to me for trying to find the root cause of this issue.

Firstly, i will understand the document on the interaction between regions and codecs on a sip gateway  written by aokanlawon, i think it is a good start.

I will notice all with updates with regard to this, i hope to find a solution soon.

Regards,

Alexis

[alemunmo81]

Hello

I have a doubt, how can i get these logs using CUCM SDI and SDL traces with SIP call processing and SIP Stack options enabled??...should i use the real time monitoring tool?

I appreciate the help.

Thanks,

Alexis