Outgoing calls not happening on SIP trunk

[Mohammed Asif C Y]

I have created a SIP trunk in CUCM 6.1 and created a route patter for ISD dialing, whenever i dial any ISD numbers i don't hear any ring back or dial tone, and after some time the call gets disconnected. SIP trunk has a public IP towards my service provider. I have check point firewall which has this public ip and opened all the UDP ports and also the SIP port 5060.

I pulled out SDI and SDL logs and found the below error

(102) Call terminated when timer expired; a recovery routine executed to recover from the error

and in the log details i got the following information

07/27/2012 16:44:03.046 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 202.x.x.x:[5060]:

INVITE sip:00xxxxxxxxxx@202.x.x.x:5060 SIP/2.0

Date: Fri, 27 Jul 2012 11:14:03 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "user" <sip:xxxx@x.x.x.x>;tag=97f1d207-9eab-4d8a-971f-ae6ac52b9e42-27002134

Allow-Events: presence

Supported: timer,replaces

Min-SE:  1800

Remote-Party-ID: "user" <sip:xxxx@x.x.x.x>;party=calling;screen=yes;privacy=off

Content-Length: 212

User-Agent: Cisco-CUCM6.1

To: <sip:00xxxxxxxxxx@202.x.x.x>

Contact: <sip:xxxx@x.x.x.x:5060>

Expires: 180

Content-Type: application/sdp

Call-ID: 2a41be80-12177fa-2ae77de-56417ac@172.23.100.5

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK159b6c1242ee4a6

CSeq: 101 INVITE

Session-Expires:  1800

Max-Forwards: 70

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 x.x.x.x

s=SIP Call

c=IN IP4 x.x.x.x

t=0 0

m=audio 25806 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Please help me resolving this and let me know what do i have to get this fixed

[Inder Manjit Singh · Accepted solution]

Hello Mohammed,

You may be running into an issue with NAT at your perimeter firewall.  You could try using a Cisco Unified Boarder Element (CUBE) to resolve this issue.  The following application note may help you to set up a CUBE:

http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/1106750.pdf

Please rate this post if it is helpful to you.

Regards.  Inder.

[Leonardo Santana]

I agree with Inder, and is more secure to use a CUBE.

You can configure address hiding, trust list and other things. Check the documentation.

Regards

Leonardo Santana

[Mohammed Asif C Y]

I have attached the wireshark logs, also i checked on NATTING part everything looks ok. But still the problem persists, as per the service provider they are getting CUCM trigger on there gateway.

[Mohammed Asif C Y]

Still No Luck on this issue, i have captured packet traces from the CLI of CUCM and gone through it from Wireshark, i am getting a traffic from my firewall and whenever it hits the CUCM IP it shows error 404 not found.

[Jayaraja Rajamani]

HellO !

I would like to know about the following check list.

A) Did you add domain name in the router ?  Is yes. Please go and add domain name in the Gateway and trunk settings?

like : Nameoftherouter.domainname

B) R u getting incoming from from all pattern ? ex: ,local, STD,ISD'

    If yes. Double check did you configgure pattern with correct predigit dot seltion at the pattern level /

    If yes. Double check ? Did your call router configure outgoing pattern and dial rule/

   Sample. If in INDIA

dial-peer voice 8000 voip

destination-pattern  (DID) $

session protocol sipv2

session target ipv4:callmanager IP

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

no vad dial-peer voice 8000 voip
destination-pattern DID$
session protocol sipv2
session target ipv4:Call manager IP

incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad

More support open case in cisco ?

Thanks

Best Regards

Jayaraja T

[Mohammed Asif C Y]

Hi Everyone,

This issue has been resolved, i just added SIP trunk in our new MCS server which has 8.6 CallManager version and we are able to make ISD calls for US and Canada, i wonder why this has not supported in 6.1 CallManager. May be some bugs related to SIP behaviour. Also in 8.6 we have an option for enabling SIP early offer settings, may be this also one of the reason. Where as 6.1 is not having any settings for SIP early offer and delay offer.

Thanks for your support

Regards

Asif C Y