CUCM - P-Asserted-Identity
Topology: CUCM ---- SIP ---- CUBE ---- SIP ---- SP
Sometimes, when I place a call, it redirects it to an internal extension. The only SIP message that I can see being sent when that happens is a 180 ringing with a p-asserted-identity in it. It only happens about 50% of the time. example: I dial 9.8001234567 and end up talking to someone at extension 3041. The Dialed Number Analyzer shows everything correctly. This SIP message is below. HELP!! please
10/01/2013 08:59:39.274 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to -CUBE--->10.3.20.240:[5060]:
SIP/2.0 180 Ringing
Date: Tue, 01 Oct 2013 13:59:39 GMT
Call-Info: <sip:-CUCM--->10.3.20.10:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: <sip:5141234567@--SP Edge device----->22.22.22.22>;tag=739821B4-19C2
Allow-Events: presence
P-Asserted-Identity: "Switchboard" <sip:[email protected]>
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Remote-Party-ID: "Switchboard" <sip:[email protected]>;party=called;screen=yes;privacy=off
Content-Length: 0
To: <sip:[email protected]>;tag=87d22a15-fd7d-492e-85ed-0dc8d67386d5-30912190
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.3.20.240:5060;branch=z9hG4bK2D01637
CSeq: 101 INVITE
My guess is that the call got hairpinned back to CUCM by the provider or CUBE. We need to see the full trace from CUBE to be sure. PAI is really just attempting to indicate who the call is actually alerting because the From and To headers cannot change after the INVITE message.
The fact that CUCM is sending a 180 RINGING message implies that it is processing an *incoming* call. This is reinforced by the headers because the From header is the SP SBC, the via header is CUBE, and the To header is CUCM. Inbound call!
Compare the Call-ID of the SIP INVITE CUCM first sends to CUBE for your outbound call to the one shown in this message. Are they different?
Run these if you want us to look deeper. If you have multiple calls going be certain to point out the calling/called number and the IPs if any differ from what you have called out above.
show run | section dial-peerdebug ccsip messagesdebug voip dialpeer
Please remember to rate helpful responses and identify helpful or correct answers.
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INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850 disconnect cause value of 21, which represents ‘Call Rejected’. The debug voip ccapi inout command can be run to identify the cause value.
Additionally, voice iec syslog can be enabled to further verify if the call failure is a result of the toll-fraud prevention. This configuration, which is often handy to troubleshoot the origin of failure from a gateway perspective, will print out that the call is being rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice gateways before this trusted address toll-fraud prevention feature was implemented. All of these configurations require that you are already running 15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be defined. This below configuration accepts calls from those host 203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from all other hosts are rejected. This is the recommended method from a voice security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone -
CUCM and Avaya CS1000 SIP connection
Hello - looking for some help on a SIP trunk configuration between the 2 devices. Currently we are running CUCM 9.1 with Avaya Session Manager 6.3. We are having issues with the call completing from the CS 1000 to the CUCM. Below are the session traces from both the Session Manager and CUCM. The CS1000 currently has 4 digit extensions and the CUCM has 7 digit extensions. We translate the number in the Session Manager to send the 7 digits. If you could lead in the right direction I would appreciate it. WOuld this have something to do with the context coming out of Session Manager? It looks like CDP.UDP and then only the 4 digits and the CUCM needs the 7. I also attached the configurations guide used for this.
Session Manager trace:
mil-ss-01 CUCM 9
SM100 10.101.2.75 10.174.2.75
13:04:53.730 |<--OPTIONS-| | | | | (1) sip:10.5.1.30
13:04:53.731 |--200 OK-->| | | | | (1) 200 OK (OPTIONS)
13:04:53,734 | Request Adaptation | Adapter: mil-ss-01
13:04:53,734 | Request Adaptation | Adapter: mil-ss-01
13:04:56.183 |--INVITE-->| | | | | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7317;phone-context=cdp.udp
13:04:56.184 |<--Trying--| | | | | (2) 100 Trying
13:04:56,185 | Remote host is trusted | Trusted
13:04:56,185 | Request Adaptation | Adapter: mil-ss-01
13:04:56,186 | Applied ingress Adaptation | P-Asserted-Identity=<sip:[email protected]>, Request-URI=sip:[email protected], History-Info=<sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
13:04:56,186 | Originating Location found | Location: mil-cs1000m-01
13:04:56,186 | Try routing to determine if emergency call | Location: mil-cs1000m-01
13:04:56,186 | Request Dial Pattern route | for: sip:[email protected] Location: mil-cs1000m-01
13:04:56,186 | Dial Pattern route parameters | URI Domain: company.com Location: mil-cs1000m-01
13:04:56,186 | Trying Dial Pattern route | Domain: company.com Location: mil-cs1000m-01
13:04:56,186 | Dial Pattern found | for: 7317 Pattern: 7317
13:04:56,186 | Route Policy found | Pattern: 7317 RoutePolicyList: to_CUCM9
13:04:56,187 | Route found | for: sip:[email protected] SIPEntity: CUCM 9
13:04:56,187 | Entity Link found | SIPEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP, biDirId=null, deny=false:5060
13:04:56,187 | Request Adaptation | Adapter: CUCM 9
13:04:56,188 | Applied egress Adaptation | NoAdaptationModuleExists=true, Request-URI=sip:[email protected];routeinfo=0-0, Remote-Party-ID=<sip:[email protected]>;party=calling;screen=no;privacy=off,
13:04:56,188 | Routing SIP request | SipEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP:5060
13:04:56,189 | No hostname resolution required | Routing to: sip:10.5.131.12;transport=tcp;lr;phase=terminating
13:04:56.191 | |--INVITE-->| | | | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7657317 P:terminating
13:04:56.196 | |<--Trying--| | | | (2) 100 Trying
13:04:56.198 | |<--Not Fou-| | | | (2) 404 Not Found
13:04:56.199 | |----ACK--->| | | | (2) sip:[email protected]
13:04:56,200 | Request Adaptation | Adapter: CUCM 9
13:04:56,201 | Request Adaptation | Adapter: CUCM 9
13:04:56,201 | Request Adaptation | Adapter: mil-ss-01
13:04:56.202 |<--Not Fou-| | | | | (2) 404 Not Found
13:04:56.203 |----ACK--->| | | | | (2) sip:7317
13:05:07,597 | Remote host is not trusted | Host not trusted
13:05:07,597 | Originating Location found | Location: mil-cs1000m-01
13:05:12.657 | |<--------OPTIONS-------| | | (3) sip:10.5.2.51
13:05:12,659 | Remote host is not trusted | Host not trusted
13:05:12,659 | Originating Location found | Location: mil-cs1000m-01
13:05:12.660 | |--------200 OK-------->| | | (3) 200 OK (OPTIONS)
13:05:16.877 | |<--------------OPTIONS-------------| | (4) sip:10.5.2.51
13:05:16,879 | Remote host is trusted | Trusted
13:05:16,879 | Request Adaptation | Adapter: mil-ss-01
13:05:16,879 | Applied ingress Adaptation | P-Asserted-Identity=<sip:[email protected]>
13:05:16,879 | Originating Location found | Location: sas-cs1000e-01
13:05:16.880 | |--------------200 OK-------------->| | (4) 200 OK (OPTIONS)
13:05:24.463 | |<--------------------OPTIONS-------------------| (5) sip:10.5.2.51
13:05:24,465 | Remote host is not trusted | Host not trusted
13:05:24,465 | Originating Location found | Location: mil-cs1000m-01
13:05:24.466 | |--------------------200 OK-------------------->| (5) 200 OK (OPTIONS)
CUCM Trace Invite:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 3188 bytes:
[1179,NET]
INVITE sip:[email protected] SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
History-Info: <sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Contact: <sip:00000000;[email protected]:5060;maddr=10.5.1.30;transport=tcp;user=phone;gsid=68cac530-5d21-11e3-8b45-78e3b505dc88>
Alert-Info: <cid:[email protected]>
Supported: 100rel, x-nortel-sipvc, replaces
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
Via: SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140
Via: SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
Record-Route: <sip:[email protected];transport=tcp;lr>
Record-Route: <sip:10.5.2.50:15060;transport=tcp;ibmsid=local.1372169047609_2400497_2400521;lr>
Record-Route: <sip:[email protected];transport=tcp;lr>
P-Charging-Vector: icid-value="68cac530-5d21-11e3-8b45-78e3b505dc88"
User-Agent: Nortel CS1000 SIP GW release_7.0 version_linux-6.50.00 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: <sip:[email protected]>
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
Max-Forwards: 66
CSeq: 1 INVITE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 1063
Av-Global-Session-ID: 68cac530-5d21-11e3-8b45-78e3b505dc88
P-Location: SM;origlocname="mil-cs1000m-01";origsiglocname="mil-cs1000m-01";origmedialocname="mil-cs1000m-01";termlocname="Cisco BE6K";termsiglocname="Cisco BE6K";smaccounting="true"
--unique-boundary-1
Content-Type: application/sdp
SDP Message
====================================================
v=0
o=- 746 1 IN IP4 10.5.1.30
s=-
c=IN IP4 10.5.1.36
t=0 0
m=audio 5234 RTP/AVP 18 0 8 101 111
c=IN IP4 10.5.1.36
a=tcap:1 RTP/SAVP
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=pcfg:1 t=1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
0500bc05
0107130081900000a200
09090f00e9a4830001004000
1315070011fa0f00a10d02010102020100cc040000c56000
1e0403008183
4a1c0100180001001a011404000067353505000004000000000048710000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
011201
3c:4a:92:f4:84:f4
--unique-boundary-1--
CUCM Trying Message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
CUCM not found message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
CUCM ACK message:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Thanks.This document worked for us between CUCM BE6000 ver 9.0 and the Avaya.
The main focus on the Cisco side is this: Page 37 - 41
5.4. Define SIP Trunk Security Profile
Expand System Security Profile and select SIP Trunk Security Profile. Click to
configure a SIP Trunk Security Profile.
Enter the following values and use defaults for remaining fields:
Name Enter name
Description Enter a brief description
Incoming Transport Type Verify “TCP+UDP” is selected
Outgoing Transport Type Verify “TCP” is selected
Accept Out-of-Dialog REFER Enter
Accept Unsolicited Notification Enter
Accept Replaces Header Enter
Click . The screen below shows SIP Trunk Security Profile for the sample configuration
5.5. Define SIP Profile
Expand Device Device Settings and select SIP Profile. Click to configure a SIP
Profile.
Under SIP Profile Information section, enter the following values and use defaults for
remaining fields:
Name Enter name
Description Enter a brief description
Default MTP Telephony Event Payload Type Enter “120”
Disable Early Media on 180 Enter
Note: Disabling Early Media allows local ringback to be used.
Under Parameters used in Phone section, scroll to end of section and enter the following values
and use defaults for remaining fields:
RFC 2543 Hold Enter
Click . The screen below shows SIP Profile for the sample configuration. -
Mitel System With Trunk To CUCM DTMF Not Recognized.
Hello,
Not sure if this is correct forum but maybe someone has seen this. Overview is that DTMF is not being recognized by CVP from the greeting. Call flow is such.
Mitel 5000 --siptrunk---CUCM9---siptrunk---CVP9
Currently the DTMF type on all trunks is RFC2833. And verified the Mitel is RFC as well. I think the problem is that for payload Mitel is sending 96. But on the Cisco CUCM the trunks are using the default of 101 for both the trunk to Mitel and CVP. Here is the invite in from Mitel you can see it sends payload of 96.
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
Max-Forwards: 70
Allow: NOTIFY,REGISTER,REFER,SUBSCRIBE,INVITE,ACK,OPTIONS,CANCEL,BYE
User-Agent: Mitel-5000-ICP-5.1.0.56
P-Asserted-Identity: "Laura B." <sip:[email protected]>
From: "Laura B." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
To: 72799 <sip:[email protected]:5060>
Call-ID: 2312043591-10065
CSeq: 1 INVITE
Contact: "Laura B." <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 266
v=0
o=Mitel-5000-ICP 169457268 1404250460 IN IP4 10.254.20.31
s=SIP Call
c=IN IP4 10.254.20.31
t=0 0
m=audio 6770 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:30
a=cdsc:1 image udptl t38
5:16 PM
here is 200 ok
5:16 PM
5:16 PM
Jay Schulze:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
From: "." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
To: 72799 <sip:[email protected]:5060>;tag=10585346~dfbf10b3-6c69-4443-852f-cbf609935a6f-42551743
Date: Tue, 01 Jul 2014 21:34:24 GMT
Call-ID: 2312043591-10065
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Preferred-Identity: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 238
v=0
o=CiscoSystemsCCM-SIP 10585346 1 IN IP4 10.38.246.136
s=SIP Call
c=IN IP4 10.38.246.166
b=TIAS:64000
b=AS:64
t=0 0
m=audio 23956 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
This is the invite over to CVP. Still sending 96. I'm pretty sure I can fix this by setting the payload type to 96 on both trunks on CUCM. I was able to test with it afterhours. After reading through Mitel docs there is no way to change payload type on their side. The question is really would there be anything else this could maybe break on the CVP side?Well that did not work. What I found was it was going through 96 all the way to CVP. However when CVP returned a label to the VXML. The VXML sent back the 200 OK with 101.
If I changed the the rtp payload-type nte 96 on the VXML dial-peer it did work. However it broke any other call flow to CVP from being able to recognize DTMF.
I wonder if this is some type of bug on VXML. Because it was my understanding by having the command 'asymmetric payload full' under SIP. It would pass the payload type from one call leg to the other. -
Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working
Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vadHi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 716 717 19314 4000 192.168.1.10 192.168.1.241
2 717 716 19234 54932 10.249.13.130 10.224.42.72
Found 2 active RTP connections -
Video endpoint registerd to VCSc calling CUCM
I have an EX90 registered to a VCSc with a SIP trunk connected to CUCM. An extension on CUCM can call the endpoint registered to the VCSc, but when a video endpoint tries to call an extension on the CUCM the call fails, but the video endpoint can dial 9+10 digitit numberaor 9+1+!0 digits. i have set up a specific search rule to match an exact extension number in CUCM to match but the call still fails. In the call history of the VCSc the result is 404 not found - I am using SIP.
HI Michael. Are you aslo using the CUBE in the same manner here also? Reading the case it seems the call has to go out via PSTN and manages to go through a CUBE. The CUBE is rejecting it because it has no valid caller id? So I'm assuming that the PAI would need to be modified in this instance when the call is passed from VCS to CUCM.
VCS can't manipulate that far down in the initial INVITE. I'd have to check to see if a CPL would do this, but I'm doubting it, but will check around for you.
if you have endpoints running TC6.3, you can ultimately try this to see if this will work for you.
1) H323 protocol on endpoint - Remove the h323ID from the system.
2) Verify E164 alias is programmed on the system and is registered to VCS.
3) Place call as H323 so that the interwork function (IWF) send call as SIP towards CUCM. Since the h323ID is missing, IWF may pick up E164 alias in P-Asserted Identity (PAI) and send the call to CUCM.
Now, you can test this...I can't guarantee it will work, but from reading the notes, they want something numeric versus alphanumeric and maybe if we place the E164 alias in the PAI it will help. Can't guarantee it though...let us know..
Thanks.
VR
P2 -
Hello,
I have a problem while using two identity asserters with two differents tokens.
My problem is:
I have two specific identity asserters i developped.
They are both in REQUIRED mode.
Identity Asserter one (tokenOne)
Identity Asserter two (tokenTwo)
I would like to fire the two asserters when the two tokens are in the request.
When the two tokens are in the request, il would like to have the name contained in the token two in the name of the request's principal.
Currently, only the first asserter is fired and the name i find in the request is the one who is contained in tokenOne.
Thanks you for your answers.
(Excuse me for my weak english)
Edited by: user5451975 on 21 janv. 2010 04:51Yes but if i return null in the method getAssertionModuleConfiguration, i have a error returned by weblogic:
javax.security.auth.login.LoginException: [Security:090300]Identity Assertion
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <PrincipalAuthenticator.assertIdentity - Token Type: ICTicket>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP ATN LoginModule initialized>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP Atn Login>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP Atn Login username: test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <userExists? user:test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <getConnection return conn:LDAPConnection { ldapVersion:2 bindDN:""}>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <getDNForUser search("ou=people,ou=myrealm,dc=devdomain", "(&(uid=test11)(objectclass=person))", base DN & below)>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <DN for user test11: uid=test11,ou=people,ou=myrealm,dc=devdomain>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <user exists, user:test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <returnConnection conn:LDAPConnection { ldapVersion:2 bindDN:""}>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP Atn Asserted Identity for test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <List groups that member: test11 belongs to>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <getConnection return conn:LDAPConnection { ldapVersion:2 bindDN:""}>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <getDNForUser search("ou=people,ou=myrealm,dc=devdomain", "(&(uid=test11)(objectclass=person))", base DN & below)>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <DN for user test11: uid=test11,ou=people,ou=myrealm,dc=devdomain>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <search("ou=groups,ou=myrealm,dc=devdomain", "(&(uniquemember=uid=test11,ou=people,ou=myrealm,dc=devdomain)(objectclass=groupOfUniqueNames))", base DN & below)>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Result has more elements: false>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <returnConnection conn:LDAPConnection { ldapVersion:2 bindDN:""}>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <login succeeded for username test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP Atn Commit>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <LDAP Atn Principals Added>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Signed WLS principal test11>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <PrincipalAuthenticator.validateIdentity>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Validate WLS principal test11 returns true>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <RoleManager.getRoles subject: Subject: 1
Principal = class weblogic.security.principal.WLSUserImpl("test11")
Resource: type=<url>, application=Cherika, contextPath=/remoting, uri=/remoteService.do, httpMethod=POST>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Default RoleMapper getRoles(): input arguments:
Subject: 1
Principal = class weblogic.security.principal.WLSUserImpl("test11")
Resource: type=<url>, application=Cherika, contextPath=/remoting, uri=/remoteService.do, httpMethod=POST>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Default RoleMapper getRoles(): returning roles: Anonymous>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <RoleManager.getRoles Subject: Subject: 1
Principal = class weblogic.security.principal.WLSUserImpl("test11")
Resource: <url> type=<url>, application=Cherika, contextPath=/remoting, uri=/remoteService.do, httpMethod=POST Anonymous roles.>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Default Authorization isAccessAllowed(): input arguments:>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < Subject: 1
Principal = class weblogic.security.principal.WLSUserImpl("test11")
>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < Roles:Anonymous>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < Resource: type=<url>, application=Cherika, contextPath=/remoting, uri=/remoteService.do, httpMethod=POST>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < Direction: ONCE>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < Context Handler: >
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> < non-null>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <Default Authorization isAccessAllowed(): returning DENY>
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <DefaultAdjudicatorImpl.adjudicate results: DENY >
####<27 janv. 2010 16 h 29 CET> <Debug> <SecurityDebug> <A82S002292> <myserver> <ExecuteThread: '13' for queue: 'weblogic.kernel.Default'> <<WLS Kernel>> <> <000000> <AuthorizationManager.isAccessAllowed returning adjudicated: false>
Why does he try to authenticate the user via LDAP ??
Edited by: user5451975 on 27 janv. 2010 07:34 -
What is the recommended way to access a database from an identity assertion provider?
For assert identity the caller should have admin privs. Depending on how the client
obtains and communicate with the EJB you have the option to setup a <run-as> tag
then map the role to an admin user via <run-as-role-assignment> using the deployment
descriptors.
-Craig
"Claude" <[email protected]> wrote:
>
Thank you for your answer !
I had not thought of building a custom "UsernamePassword" authenticator
and use
the password as token. It seams to be a good simple way of solving the
problem.
The EJB that accepts a token as parameter is interesting as well (especially
if
they are
performance issues). Are you sure that the class
weblogic.security.services.Authentication is suitable to be called from
an EJB?
Thanks, Claude
"Craig" <[email protected]> wrote:
I don't know of many uses outside of WebService or WebApp except when
using 2-way
SSL where the client certificate can be used to assert identity.
You could write an EJB that accepted the token and then the EJB would
programmatically
call identity assertion. Or if you had a custom authenticator you could
take the
token as the "password".
http://edocs.bea.com/wls/docs81/javadocs/weblogic/security/services/Authentication.html
-Craig
"Claude" <[email protected]> wrote:
Hello
I'm wondering whether an Identity Assertion Provider can be used with
a Java-client
or
if they are only for Web-clients (or Web-services).
I'm also wondering how the client sends its token (through the credential
set
of the JAAS
subject ?).
Thanks
Claude -
SIP Conversion - No longer able to make modem calls
We are working to convert our voice network off of PRI's and on to SIP Trunks.
Call Flow:
ITSP—SIP—CUBE—H323—CUCM—H323—Branch Router(2911 or 2811)—Analog Device (Cash Advance Machine) on FXS port.
All locations that have converted to SIP are now unable to connect to the analog toll free number that the analop Cash Advance Machine Dials. Getting the following error: SIP/2.0 488 Not Acceptable Media. Debugs are below for CUBE device. We have the dial peers set at the branch level to send G.711. Within CUCM we have the region settings at
Factory Default lossy
64 kbps (G.722, G.711)
384
As you cas see in the logs it is still coming in as below and should probably be G.711:
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
Can anyone suggest why the codec is being compressed? I assume this is why the call setup is failing.
Thanks for the help!
**I have also attached a file for CUCM debug. Calling Number = 317-596-8401 Called Number = 800-741-3737
Here are some debugs ran:
Feb 3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:[email protected]>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:[email protected]>
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2884739328-0000065536-0000081835-0193008300
Session-Expires: 1800
P-Asserted-Identity: "7652847084" <sip:[email protected]>
Remote-Party-ID: "7652847084" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 275
v=0
o=CiscoSystemsCCM-SIP 873207 1 IN IP4 172.1.1.1
s=SIP Call
c=IN IP4 10.200.130.53
b=TIAS:8000
b=AS:8
t=0 0
m=audio 18874 RTP/AVP 18 4 101
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Feb 3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:[email protected]>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:[email protected]=65597134-9EC
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 304 32.2XX.X.XX "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Feb 3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:[email protected]>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:[email protected]>;tag=65597134-9EC
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
Message was edited by: Jake RieckenIn Skype open Tools -> Options -> Advanced settings. At the bottom of this screen you should see a menu item saying “Manage other programs access to Skype”. Click on this menu. Do you see any applications listed there?
-
Unified Mobility, no audio on Send call to Mobile Phone
I'm using UCM 9.1 with Unified Mobility (xfer to alternate number) with good success if I follow a typical call flow:
Inbound call -> Ext -> Rings deskphone x seconds -> Rings mobile phone -> Answer mobile phone -> hangup -> Resume call on desk phone.
But if I pick up a call on my desk phone, and use the Mobility button to xfer a call to my mobile phone I get no audio:
Inbound call -> Ext -> Pickup desk phone -> Mobility soft button, 'Send call to Mobile Phone' -> Answer mobile phone, no audio -> Hang up mobile phone -> Resume call on desk phone (two-way audio).
Device wise the call flow is:
ITSP SIP trunk -> CUBE -> CUCM -> 7965 IP Phone.
Recently I reconfigured CUCM to use the CUBE for any MTP resources instead of the software option and I think I may have missed something.
CUBE config:
voice-card 0 dspfarm dsp services dspfarm!!!voice service voip ip address trusted list ipv4 173.46.30.218 ipv4 173.46.30.202 ipv4 10.0.6.30 ipv4 10.0.6.31 ipv4 10.0.6.33 ipv4 10.0.6.32 ipv4 10.1.1.4 ipv4 10.0.250.0 255.255.255.0 mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip midcall-signaling passthru media-change early-offer forced no call service stop registration passthrough!voice class codec 1 codec preference 1 g711ulawsccp local GigabitEthernet0/0.42sccp ccm 10.0.6.30 identifier 1 version 7.0 sccp!sccp ccm group 1 bind interface GigabitEthernet0/0.42 associate ccm 1 priority 1 associate profile 1 register MTP_2951-01!dspfarm profile 2 transcode universal codec pass-through codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 11 associate application SCCP shutdown!dspfarm profile 3 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP!dspfarm profile 1 mtp codec pass-through codec g711ulaw maximum sessions hardware 15 associate application SCCP!
In CUCM I removed the software MTP from the MRG:
Not sure where to start troubleshooting this problem, any help is appreciated.
SteveThanks for the help gents. I couldn't get to this until we're out of office hours on the weekend.
Interestingly, I have no mid-call option in my dial-peers. This is a 2951 running 15.2(4)M2.
I double checked the MRGL, my phone is associated with it.
Codec is G711 on ITSP side, and on phones - I'm not sure I fully understand the use cases for MTP, this is something I need to research more.
I've included two ccsip message debugs, the first one is the existing issue of no audio (in either direction).
The second I've changed the midcall-signaling passthru option, dropping the media-change bit and we get audio in both directions for mobility except we use Unity call handlers for IVR functionality, and now when an inbound caller is forwarded to an extension we get no audio - obviously this is a game stopper.
In Unity I have the port group configured as SCCP - Maybe I should be using SIP instead?
No Audio:
voice service voip
sip
midcall-signaling passthru media-change
early-offer forced
no call service stop
registration passthrough
Dec 8 21:07:35.842: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 3244526336-0000065536-0000003600-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:07:35.850: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 3244526336-0000065536-0000003600-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536855Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9952 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:35.850: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:35.858: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Dec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120698 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:43.938: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120699 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:43.938: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:43.942: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274C4F2From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:07:43.958: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1d288c5e53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40265 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:44.158: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.158: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:07:44.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationSession-Expires: 1800;refresher=uacP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:44.214: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2151dd5c40From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40265 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 25834 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:52.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120700 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9953 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:52.610: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK8m4hbg10c8ag4kg723g0.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:07:52.630: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:07:52.634: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386536872CSeq: 101 BYEReason: Q.850;cause=16P-RTP-Stat: PS=0,OS=0,PR=420,OR=67200,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.634: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=2,OS=320,PR=100,OR=16000,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.646: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 101 BYEContent-Length: 0
bi-direcitonal audio:
voice service voip
sip
early-offer forced
midcall-signaling passthru
no call service stop
registration passthrough
Dec 8 21:09:44.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 0239559040-0000065536-0000003601-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:09:44.339: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536984Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesSession-Expires: 1800Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2507 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:44.339: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:44.347: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536984Dec 8 21:09:52.535: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674502 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:52.539: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:53.007: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:09:53.007: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674503 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:54.967: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274F1060From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:09:54.979: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2876c28ab9From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40276 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.007: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:09:55.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 102 INVITEMax-Forwards: 70Timestamp: 1386536995Contact: Diversion: ;privacy=off;reason=follow-me;screen=yesExpires: 180Allow-Events: telephone-eventContent-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:55.019: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Dec 8 21:09:55.031: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 279v=0o=root 1961674502 1961674504 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:55.035: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:55.175: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2c8e71c96From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.179: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK2751A88From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 102 ACKAllow-Events: telephone-eventContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674505 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:10:05.271: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITEMax-Forwards: 70Timestamp: 1386537005Contact: Expires: 180Allow-Events: telephone-eventContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.271: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceSupported: replacesSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=called;screen=yes;privacy=offContact: Content-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:10:05.279: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.279: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27531E8DFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4d5qq910785h6ks9f3c0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:10:05.295: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386537005CSeq: 102 BYEReason: Q.850;cause=16P-RTP-Stat: PS=511,OS=81760,PR=505,OR=80800,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=505,OS=80800,PR=634,OR=101440,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.303: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 BYEContent-Length: 0
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