[SR-Users] Multi-homed Kamailio Modifies ACK
SamyGo
govoiper at gmail.com
Tue May 12 18:24:16 CEST 2015
Yeah sure, attached here is the trace from the CISCO logs. Not in pcap
format.
Also this is the flow of the call.
+-----------+ +--------------+ +------------------+
|Provider |+--------> +Kamailio +------>+CISCO IAD |
+-----------+ +--------------+ +-----------------+
| ^
v |
+-----------+
|Asterisk |
+-----------+
Thanks and best Regards,
--
On Tue, May 12, 2015 at 11:50 AM, Alex Balashov <abalashov at evaristesys.com>
wrote:
> Is the UAC in this case a 2543 endpoint by chance? Can you send a complete
> SIP trace of the entire setup?
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
> *From: *SamyGo
> *Sent: *Tuesday, May 12, 2015 11:46
> *To: *SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Reply To: *Kamailio (SER) - Users Mailing List
> *Subject: *[SR-Users] Multi-homed Kamailio Modifies ACK
>
> Hi all,
>
> I'm having a scenario where I'm sending call to CISCO IAD2431. The call
> establishes with 200OK from CISCO and Kamailio sends back modified ACK.
> Cisco at this points gives out an error and sends 200OK again, Kamailio
> replies with ACK, and this cycle goes on until the call drops.
>
> Here is the error from CISCO:
>
> Failed FROM/TO Request check
> old_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
> old_to: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
> new_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
> new_to: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FE
>
> I've
> #auto_aliases=no
>
> and,
> listen=udp:44.33.22.11:5041
> listen=udp:192.168.1.244:5041
>
> Kamailio sends INVITE through the Public IP to the CISCO gw, but decides
> to change the ACK header.
>
> Any advise please.
>
> Best Regards,
> Sammy.
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.05.11 15:38:15 =~=~=~=~=~=~=~=~=~=~=~=
CISCO-GW#
CISCO-GW#
CISCO-GW#t show log
006271: *May 12 04:26:06.939: Received:
INVITE sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Record-Route: <sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1
Date: Mon, 11 May 2015 19:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-TRUNKPORT: 5060
P-TRUNKIP: 99.110.111.112
P-UDOMAIN: sip.iamip.com
Content-Type: application/sdp
Content-Length: 301
v=0
o=root 1791803783 1791803783 IN IP4 44.33.22.12
s=Asterisk PBX 11.13.1
c=IN IP4 44.33.22.12
t=0 0
m=audio 51590 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
006329: *May 12 04:26:06.995: Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
006380: *May 12 04:26:07.027: Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006404: *May 12 04:26:10.535: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006405: *May 12 04:26:10.547: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.4509eb49528e4c78295970355b27eea0.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK117c2473;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006423: *May 12 04:26:11.039: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006424: *May 12 04:26:11.087: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.c61fe7a84dd9b53ca1913f670a35974c.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK4c9a6eec;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006425: *May 12 04:26:11.091: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 44.33.22.11:5041
006426: *May 12 04:26:11.091: *****CCB found in UAS Request table. ccb=0x03343698
006427: *May 12 04:26:11.095: Failed FROM/TO Request check
old_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
old_to: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
new_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
new_to: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
006428: *May 12 04:26:11.095: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg
006429: *May 12 04:26:11.095: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg : Invalid method for this state (STATE_IDLE): ACK
006430: *May 12 04:26:12.027: CCSIP-SPI-CONTROL: act_sentsucc_wait_ack
006431: *May 12 04:26:12.027: CCSIP-SPI-CONTROL: sipSPIIncomingCallSDP
006432: *May 12 04:26:12.027: SDP already there use old sdp and updatemedia if needed
006442: *May 12 04:26:12.035: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006443: *May 12 04:26:12.051: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.59b9952218c019d3d5270ba9ab05a754.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK4058d3b8;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006476: *May 12 04:26:14.039: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006477: *May 12 04:26:14.059: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.05a987bb7ea3d860140ae86a3ed4e416.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK75fe11db;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006497: *May 12 04:26:17.063: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.30.96.194:9595;branch=z9hG4bK2ec1860d
From: "Unknown" <sip:Unknown at 4.30.96.194:9595>;tag=as3bb45eab
To: <sip:99.110.111.112>;tag=9FEC7E92-1688
Date: Sat, 28 Aug 1993 04:26:17 GMT
Call-ID: 4abe305524624906710c7ede65d8245f at 4.30.96.194:9595
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 167
v=0
o=CiscoSystemsSIP-GW-UserAgent 4275 1136 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 0 RTP/AVP 18 0 8 2 0 0
c=IN IP4 99.110.111.112
006510: *May 12 04:26:18.043: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006511: *May 12 04:26:18.059: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.c1406258ae04c82d901df89fec2a79c0.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK0e1203fd;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006512: *May 12 04:26:18.059: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 44.33.22.11:5041
006513: *May 12 04:26:18.063: *****CCB found in UAS Request table. ccb=0x03343698
006514: *May 12 04:26:18.063: Failed FROM/TO Request check
old_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
old_to: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
new_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
new_to: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
006515: *May 12 04:26:18.067: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg
006516: *May 12 04:26:18.067: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg : Invalid method for this state (STATE_IDLE): ACK
006517: *May 12 04:26:22.035: CCSIP-SPI-CONTROL: act_sentsucc_wait_ack
006518: *May 12 04:26:22.035: CCSIP-SPI-CONTROL: sipSPIIncomingCallSDP
006519: *May 12 04:26:22.035: SDP already there use old sdp and updatemedia if needed
006529: *May 12 04:26:22.047: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006530: *May 12 04:26:22.059: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.c41541fbeb7a1a2b1ae15dd5207e9ba6.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK7fd41912;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006531: *May 12 04:26:22.059: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 44.33.22.11:5041
006532: *May 12 04:26:22.063: *****CCB found in UAS Request table. ccb=0x03343698
006533: *May 12 04:26:22.063: Failed FROM/TO Request check
old_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
old_to: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
new_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
new_to: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
006534: *May 12 04:26:22.063: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg
006535: *May 12 04:26:22.067: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg : Invalid method for this state (STATE_IDLE): ACK
006536: *May 12 04:26:26.039: CCSIP-SPI-CONTROL: act_sentsucc_wait_ack
006537: *May 12 04:26:26.039: CCSIP-SPI-CONTROL: sipSPIIncomingCallSDP
006538: *May 12 04:26:26.039: SDP already there use old sdp and updatemedia if needed
006548: *May 12 04:26:26.047: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.ddb600ce64f62e3d27a4570903cfd827.0,SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2d047b8c;rport=5060
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
Date: Sat, 28 Aug 1993 04:26:06 GMT
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:+14444812211 at 99.110.111.112:5060>
Record-Route: <sip:44.33.22.11:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>,<sip:192.168.1.244:5041;r2=on;lr=on;ftag=as370915fc;did=e28.f8c1;nat=yes>
Content-Type: application/sdp
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 1226 4741 IN IP4 99.110.111.112
s=SIP Call
c=IN IP4 99.110.111.112
t=0 0
m=audio 19428 RTP/AVP 0 101
c=IN IP4 99.110.111.112
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006549: *May 12 04:26:26.063: Received:
ACK sip:+14444812211 at 99.110.111.112:5060 SIP/2.0
Via: SIP/2.0/UDP 44.33.22.11:5041;branch=z9hG4bKff67.52ee66c1476a51da71aa42b212f2df5e.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK376e1031;rport=5060
Max-Forwards: 69
From: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
To: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
Contact: <sip:+14432232221 at 192.168.1.106:5060>
Call-ID: 29d42683248faec624fb0ce94e04ebf9 at 192.168.1.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0
006550: *May 12 04:26:26.063: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 44.33.22.11:5041
006551: *May 12 04:26:26.067: *****CCB found in UAS Request table. ccb=0x03343698
006552: *May 12 04:26:26.067: Failed FROM/TO Request check
old_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
old_to: <sip:+14444812211 at sip.iamip.com:5041>;tag=9FEC572E-100F
new_from: "Test Phone" <sip:+14432232221 at 192.168.1.106>;tag=as370915fc
new_to: <sip:+14444812211 at 192.168.1.244:5041>;tag=9FEC572E-100F
006553: *May 12 04:26:26.067: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg
006554: *May 12 04:26:26.067: CCSIP-SPI-CONTROL: sipSPISipIncomingMsg : Invalid method for this state (STATE_IDLE): ACK
006555: *May 12 04:26:30.039: CCSIP-SPI-CONTROL: act_sentsucc_wait_ack
006556: *May 12 04:26:30.039: CCSIP-SPI-CONTROL: act_sentsucc_wait_ack : Out of retries
006557: *May 12 04:26:30.039: Categorized cause:102, category:2
CISCO-GW#
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