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Modified CUBE SIP Header not matching dial-peer

p-blalock
Level 1
Level 1

I’m having a trouble with modifying my headers and getting it to work on my CUBE outbound sip profile. According to my “debug ccsip message” out put, my header modification is correct, but the inew INVITE isn’t matching my outbound dial-peers. I’m trying to copy the “Diversion” header, from an inbound call, to the “To” header; this works, the new INVITE shows the correct, copied Diversion of \<sip:1234567891@addressofCUCM;user=phone>\". I have dial-peers facing the CUCMs with destination patterns of which the new INVITE should match (ex: destination-pattern 123…….), the CUCM side is setup correctly to receive and process the digits, but they’re not being delivered by the CUBE gateway. This is driving me crazy. Why can’t I get the modified “To” header to match my CUCM facing dial-peers and deliver the new digits?

Here are the details:

response ANY sip-header Allow-Header modify "UPDATE," ""
 request INVITE sip-header Diversion copy "<sip:(.*)@(.*)>" u01
 request INVITE sip-header To modify "<sip:5551215555@(.*)>" "<sip:\u01@\1;user=phone\2>"
 request INVITE sdp-header Audio-Attribute add "a=ptime:30"
 request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
 response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"

INPUT header to modify (invite from service provider):

INVITE sip:5551215555@10.253.2.1:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKudo4390058ruj4ghlso0.1
From: "Unavailable" <sip:3233969119@x.x.x.x;user=phone>;tag=2005487715-1532903387839-
To: "User 5551215555"<sip:5551215555@10.253.2.1;user=phone>
Call-ID: BW222947839290718879647955@Q1
CSeq: 873773408 INVITE
Contact: <sip:x.x.x.x:5060;transport=udp>
P-Asserted-Identity: "Unavailable" <sip:3233969119@x.x.x.x;user=phone>
Privacy: none
Diversion: <sip:1234567891@x.x.x.x:5060;user=phone>;privacy=off;reason=no-answer;counter=1
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,NOTIFY,PRACK,OPTIONS



OUTPUT (modified header):

INVITE sip:5551215555@10.2.70.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4
From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977
To: <sip:1234567891@10.2.70.12;user=phone>
Date: Sun, 29 Jul 2018 15:29:47 GMT
Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3157893444-2461340136-3007360536-0843712768
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1532903387
Contact: <sip:3233969119@10.2.70.9:5060>
Call-Info: <sip:10.2.70.9:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 1800
Allow-Events: kpml, telephone-event
Max-Forwards: 67
P-Asserted-Identity: "Unavailable" <sip:3233969119@10.2.70.9>
Diversion: <sip:1234567891@10.2.70.9>;privacy=off;reason=no-answer;counter=1
Session-Expires:  1800;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 285

NOW TRYING:

Jul 29 22:29:47.953: //189909/BC39A144B340/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4
From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977
To: <sip:1234567891@10.2.70.12;user=phone>
Date: Sun, 29 Jul 2018 22:29:47 GMT
Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


Jul 29 22:29:47.956: //189909/BC39A144B340/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4
From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977
To: <sip:1234567891@10.2.70.12;user=phone>;tag=436422~edc4535b-0860-4aaa-b294-a1dce0d77157-40417867
Date: Sun, 29 Jul 2018 22:29:47 GMT
Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Server: Cisco-CUCM11.5
Session-ID: 00000000000000000000000000000000;remote=9096a4ce9c545503247c7b7dab436422
Content-Length: 0


DIAL PEERS - OUT TO CUCM:

dial-peer voice 4001 voip
 description Incoming Service provider - CUBE side to UCM
 preference 1
 destination-pattern 123.......
 session protocol sipv2
 session target ipv4:10.2.70.12
 voice-class codec 2  
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 no voice-class sip early-offer forced
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte sip-kpml sip-info sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad


So, what I'm I missing?  Thanks for any help/comments

2 Replies 2

Sreekanth Narayanan
Cisco Employee
Cisco Employee
The CUCM is receiving the INVITE message from the CUBE. That's why we receive the "100 Trying" and "404 Not Found" from CUCM. The CUCM is unable to route that call to the destination. It is perhaps processing the Req URI and not the To header.
Also, the cause value for disconnect is 1, so please check if you have the right values set for Significant Digits and CSS on the SIP trunk.

R0g22
Cisco Employee
Cisco Employee
First thing to check is CSS/Partition. If "5551215555" is a DN on a device, check the CSS on the SIP trunk. It should have the partition for that DN. If this number is being routed through a TP, CSS on SIP trunk should have partition of the TP and CSS of TP should have the partition for the DN/transformed number.