Hello,
I have a customer I am working with who is getting an error tranfering alls between a Cisco UCM and Nortel CS1000 across SIP Trunks via an Aaya Session Manager, when a specific condition is met. They are getting an error "SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'" We have been able to replicate this in the lab and it is a repeatable trouble
The environment is as follow
- Cisco CUCM 7.1(3)
- Two Cisco 3945 H.323 voice gateways each with a single PRI
- Aavya Session Manager
- two Nortel CS1000 PBX's
- all connected via SIP Trunks
The Call scienario that generates our error is:
Call comes in on the PRI to the main number from a blocked number (no caller name or number provided). The main number for this site a switchboard operator registered to one of the Nortel CS1000's, so there is a route pattern in place ( [0-8]XXX ) that routes this call across the SIP Trunk. The switchboard operator is able to transfer this call to another user on the same CS1000 fine, they can transfer back across the SIP Trunk to the other CS1000 fine, but if they transfer to back to a Cisco phone, they get fast busy, and appears Cisco UCM is dropping the call and providing a "SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'" response
If I place this same call from a phone that presents caller ID it works fine. it is just when call display is restricted.
As I mentioned we are able to replicate this in our lab, so we took a look at the SIP signalling going back and forth
We notice in the SIP headers that the initial call from the Cisco UCM to the CS1000 is showing the From field as "From: <sip:anonymous@anonymous.invalid;user=phone>" when the call comes back to Cisco UCM from CS1000 we see the same information int the from field, and we see a contact field of "Contact: <sip:anonymous.invalid;phone-context=UnknownUnknown@taclab.local:5060;maddr=10.240.0.1;transport=tcp;user=phone>"
Below is the signalling we are seeing on the call back to Cisco UCM from the Nortel CS1000
INVITE sip:5249@10.133.2.70;routeinfo=1-0 SIP/2.0
Record-Route: <sip:10.132.100.8:15060;lr;sap=38353230*1*016asm-callprocessing.sar140739594~1353534505117~1684854724~1;transport=tcp>
Record-Route: <sip:52545568@10.132.100.9;transport=tcp;lr>
From: <sip:anonymous@anonymous.invalid;user=phone>;tag=2165ed0-100f00a-13c4-55013-196abb-246bcc08-196abb
Call-ID: 549d290-100f00a-13c4-55013-196abb-2f0734c2-196abb
CSeq: 1 INVITE
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C0198623
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C1198621
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C1198620
Via: SIP/2.0/TCP 10.132.100.9;branch=z9hG4bK-196abb-6348edf2-7206b5f1-AP;ft=3950
Via: SIP/2.0/TCP 10.240.0.1:5060;branch=z9hG4bK-196abb-6348edf2-7206b5f1
Supported: 100rel, x-nortel-sipvc, replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.50.17 AVAYA-SM-6.1.6.0.616008
Alert-Info: <cid:external@taclab.local>
Contact: <sip:anonymous.invalid;phone-context=UnknownUnknown@taclab.local:5060;maddr=10.240.0.1;transport=tcp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 881
To: <sip:5249@taclab.local>
P-Asserted-Identity: <sip:anonymous.invalid@taclab.local>
History-Info: <sip:5249@taclab.local>;index=1,<sip:5249@taclab.local>;index=1.1
Remote-Party-ID: <sip:anonymous.invalid@taclab.local>;party=calling;screen=no;privacy=off
Route: <sip:10.132.100.9;transport=tcp;lr>
Route: <sip:10.133.2.70;transport=tcp;lr;phase=terminating>
P-AV-Transport: AP;fe=10.240.0.1:54404;ne=10.132.100.9:5060;tt=TCP;th;timerB=4
P-Location: SM;origlocname="Saskatoon";termlocname="Regina_Cisco"
Max-Forwards: 67
--unique-boundary-1
Content-Type: application/sdp
v=0
o=- 55830 1 IN IP4 10.240.0.1
s=-
c=IN IP4 10.240.10.201
t=0 0
m=audio 5200 RTP/AVP 0 8 101 111
c=IN IP4 10.240.10.201
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=maxptime:20
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=ssLinux-7.50.17;base=x2611
Content-Disposition: signal;handling=optional
0500c001
0107130081900000a200
09090f00e9a0830001004000
131e070011fd1800a1160201010201a1300e8102010582010184020000850104
131b070011fa1500a11302010102020100cc040000344bcd040000364b00
460e01000a0001000100010000000000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=ssLinux-7.50.17;base=x2611
Content-Disposition: signal;handling=optional
011201
00:1b:ba:fd:45:d7
--unique-boundary-1--
-------------------------
SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'
Reason: Q.850;cause=100
From: <sip:anonymous@anonymous.invalid;user=phone>;tag=2165ed0-100f00a-13c4-55013-196abb-246bcc08-196abb
Content-Length: 0
To: <sip:5249@taclab.local>;tag=1591439036
Call-ID: 549d290-100f00a-13c4-55013-196abb-2f0734c2-196abb
P-Av-Transport: AP;fe=10.133.2.70:5060;ne=10.132.100.9:27016;tt=TCP;th
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C0198623
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C1198621
Via: SIP/2.0/TCP 10.132.100.8:15070;branch=z9hG4bK0A846408FFFFFFFFD25CBA2C1198620
Via: SIP/2.0/TCP 10.132.100.9;branch=z9hG4bK-196abb-6348edf2-7206b5f1-AP;ft=3950
Via: SIP/2.0/TCP 10.240.0.1:5060;branch=z9hG4bK-196abb-6348edf2-7206b5f1
CSeq: 1 INVITE
-------------------------
My question is:
1> Is there something we can do on the Cisco UCM 7.1(3) to allow it to accept this call?
2> If not, we are upgrading to 8.6 in a few weeks, will we be able to address this, and allow this call to proceed using the SIP normalization scripting available in 8? If so can you provide some guidance on how to create this script.
Thank you
Denis
