[cisco-voip] CUBE Issues with SIP Reinvite...

[Jonathan Charles]

So, our carrier is routing toll free calls to a third-party and those calls
are getting a reinvite invoked.

We are ignoring the reinvite and they dump the call.

Topology is SIP Provider - SIP - CUBE - H.323 - CCM 8.6


I have added this sip-profile (from a previous post)...

voice service voip
 ip address trusted list
  ipv4 10.0.0.0 255.0.0.0
  ipv4 172.16.0.0 255.240.0.0
  ipv4 172.20.0.0 255.255.255.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol pass-through g711ulaw
 h323
  h225 display-ie ccm-compatible
 modem passthrough nse payload-type 101 codec g711ulaw
 sip
  bind control source-interface GigabitEthernet0/0.700
  bind media source-interface GigabitEthernet0/0.700
  sip-profiles 1
!
voice class sip-profiles 1
 request REINVITE sip-header Allow-Header modify "UPDATE, " " "
 request INVITE sip-header Allow-Header modify " UPDATE, " " "
 response 180 sip-header Allow-Header modify " UPDATE, " " "
 response 200 sip-header Allow-Header modify " UPDATE, " " "
!

At the 15 minute mark of the call, we get the reinvite to see if we are
still alive:


*Feb  7 17:41:29.419 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:3015552222 at 2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 72.11.192.82:5060
;branch=z9hG4bKpeibqn0080kgu05p7220sbc1rosu1.1
From: <sip:8885551111 at 72.11.192.82>;tag=266630096-1328635552106
To: <sip:3015552222 at 2.2.2.2>;tag=3B2EDDA4-E82
Call-ID: B5972919-50E711E1-A08CC205-1598096B at 2.2.2.2
CSeq: 745327741 INVITE
Contact: <sip:72.11.192.82:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Supported:
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 19
Content-Type: application/sdp
Content-Length: 179

v=0
o=- 23022424 1 IN IP4 1.1.1.1
s=-
c=IN IP4 1.1.1.1
t=0 0
m=audio 41510 RTP/AVP 0 105
a=silenceSupp:off - - - -
a=ptime:20
a=rtpmap:105 telephone-event/8000

*Feb  7 17:41:29.427 UTC: //4748/0092A0C5E201/SIP/Msg/ccsipDisplayMsg:
Sent:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.11.192.82:5060
;branch=z9hG4bKpeibqn0080kgu05p7220sbc1rosu1.1
From: <sip:8885551111 at 72.11.192.82>;tag=266630096-1328635552106
To: <sip:3015552222 at 2.2.2.2>;tag=3B2EDDA4-E82
Date: Tue, 07 Feb 2012 17:41:29 GMT
Call-ID: B5972919-50E711E1-A08CC205-1598096B at 2.2.2.2
CSeq: 745327741 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


We ignore this reinvite (CUBE thing), and the call is dumped by the remote
side.



*Feb  7 17:42:29.415 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:3015552222 at 2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 72.11.192.82:5060
;branch=z9hG4bKpeibqn0080kgu05p7220sdc1rocv1.1
From: <sip:8885551111 at 72.11.192.82>;tag=266630096-1328635552106
To: <sip:3015552222 at 2.2.2.2>;tag=3B2EDDA4-E82
Call-ID: B5972919-50E711E1-A08CC205-1598096B at 2.2.2.2
CSeq: 745327742 BYE
Max-Forwards: 19
Content-Length: 0


*Feb  7 17:42:29.435 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 72.11.192.82:5060
;branch=z9hG4bKpeibqn0080kgu05p7220sdc1rocv1.1
From: <sip:8885551111 at 72.11.192.82>;tag=266630096-1328635552106
To: <sip:3015552222 at 2.2.2.2>;tag=3B2EDDA4-E82
Date: Tue, 07 Feb 2012 17:42:29 GMT
Call-ID: B5972919-50E711E1-A08CC205-1598096B at 2.2.2.2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 745327742 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=47266,OS=7558192,PR=47707,OR=7633120,PL=0,JI=0,LA=0,DU=943
Content-Length: 0



We have spiked the SIP Session Expires Timer to 7800 seconds, but that has
had no effect... the SIP provider says they are conforming to the SIP
standard and it is us that is configured incorrectly.

Any ideas?

Jonathan
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