07-29-2012 08:50 AM - edited 03-19-2019 05:18 AM
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
Solved! Go to Solution.
07-29-2012 12:19 PM
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.
07-29-2012 12:19 PM
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.
07-29-2012 06:05 PM
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
07-30-2012 12:40 AM
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.
08-12-2012 10:36 AM
08-13-2012 03:34 AM
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
08-13-2012 12:37 PM
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
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