Outbound call failure in TelePresence Server
Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks.
Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks.
Similar Messages
-
Error messages in 2651XM GW, cause outbound call failure, reboot fix it
Cisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
Reboot fix it but there're still error messages...
How to fix it? It's IOS bug or hardware issue? How to identify?
Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Wed 03-Aug-05 20:45 by hqluong
ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
cpchn1-g1 uptime is 6 hours, 56 minutes
System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
System restarted at 03:56:27 NZST Tue Apr 17 2007
System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
Processor board ID JAE072000AJ (1555074759)
M860 processor: part number 5, mask 2
2 FastEthernet interfaces
62 Serial interfaces
2 Channelized E1/PRI ports
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
See attach detail error messagesCisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
Reboot fix it but there're still error messages...
How to fix it? It's IOS bug or hardware issue? How to identify?
Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Wed 03-Aug-05 20:45 by hqluong
ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
cpchn1-g1 uptime is 6 hours, 56 minutes
System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
System restarted at 03:56:27 NZST Tue Apr 17 2007
System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
Processor board ID JAE072000AJ (1555074759)
M860 processor: part number 5, mask 2
2 FastEthernet interfaces
62 Serial interfaces
2 Channelized E1/PRI ports
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
See attach detail error messages -
Outbound Call Failure - SIP Trunk
All phones are unable to dial a single target number on the PSTN. The symptom is that it rings once and goes fast busy.
The call flow is:
Phone >>> CUCM >>> CUBE >>> Verizon SIP Trunk >>> PSTN >>> Target Number
As seen in the CUBE debug ccsip messages, the CUBE receives a "SIP/2.0 480 Temporarily unavailable" message. debug ccsip messages, dial-peer and voice class information follows:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
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, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
Session-Expires: 1800
P-Asserted-Identity: "" <sip:[email protected]>
Remote-Party-ID: "" <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: 390
v=0
o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.171
b=TIAS:64000
b=AS:64
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387402810
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 348
v=0
o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 23372 RTP/AVP 0 8 116 18 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Supported:
Contact: <sip:[email protected]:5073;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=18
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
dial-peer voice 9100 voip
description inboubd dial-peer for outgoing calls from CUCM (11D)
preference 1
session protocol sipv2
incoming called-number ^1..........$
voice-class codec 10
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
outbound DP
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 10
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class codec 10
codec preference 1 transparent
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g722-64I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful.
See new voice class:
#sh run | be voice class codec 11
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
See revised dial-peer 8100:
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 11
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
My only remaining question is why did the CUBE invite NOT include the m line for g729r8?
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the ccapi inout snippet showing the hit with dial-peer 8100:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
From: "XXXXXXXXXX" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788
To:
Date: Thu, 19 Dec 2013 20:36:14 GMT
Call-ID: [email protected]
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, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
Session-Expires: 1800
P-Asserted-Identity: "XXXXXXXXXX"
Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 464
v=0
o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.52
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26738 RTP/AVP 0 8 116 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:116 iLBC/8000
a=ptime:30
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See ccsip messages output showing CUBE sending invite to Verizon:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
Remote-Party-ID: "David Callahan" ;party=calling;screen=yes;privacy=off
From: "David Callahan" ;tag=7DE0957C-1CAB
To:
Date: Thu, 19 Dec 2013 20:27:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387484877
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 32502 RTP/AVP 0 8 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 -
HELP!!! - EDI Outbound HTTP call failure
Our EDI outbound (HTTPS-OXTA) is failing since Monday in production. We narrowed down the area that might be an issue. This is what we see in the Apache log,
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(469): OPM: EW: Enters opm_ew_broadcast()
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(517): OPM: EW: Broadcasts msg: cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(291): OPM:hc: Connecting to url: <server_url>:8101/oprocmgr-service
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(314): OPM:hc: Connection to host: <server_url>, port: 8101
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(438): OPM:hc: HTTP Request sent to server: POST /oprocmgr-service<server_url> HTTP/1.1^M
Host: <server_url>^M
Content-Type: application/x-www-form-urlencoded^M
Content-Length: 269^M
cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[0] is HTTP/1.1 404 Not Found
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[1] is Date: Tue, 22 May 2012 05:31:56 GMT
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[2] is Transfer-Encoding: chunked
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[3] is Content-Type: text/html; charset=iso-8859-1
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(525): OPM: EW: Broadcasts to <server_url> and send result=404
I'm trying to understand the steps of the process. Does "HTTP/1.1 404 Not Found" response to the opm_hc.c(438) call? When I type "<server_url>:8101" in the browser, I get "The webpage cannot be displayed" error. Does this should work?
EDI outbound is routed to proxy and confirmed that call from OTA was never made to proxy. Switched protocol to SMTP and it worked. There is no issue other than HTTP initial call failure. Any help you can give me I'd appreciated.George great support so far (+5)
Hi Robert
debug ccsip all is very intensive so you should do the following before enabling the debug
service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
Then..
<Enable debugs, then test again.>
debug ccsip all
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
++++
What is even more strange is that the call appears to be disconnected from the far end. From the logs below the outbound call leg (45) is where the disconnect is coming from and the "cc_api_call_disconnected" shows this call leg talking to CCAPI..
001858: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x3CE6D670, Call Id=45
001859: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Can you also send a debug voip ccapi onout from the CUBE. we need to check if the call arrives there, though we don't see any INVITE request sent out. -
Outbound Calls stop working when Lync Edge server is offline
All,
We have had an issue inside our environment after one of our virtual hosts died and took out our sole Edge server, basically users could not dial out and were getting the error "Network is busy" on the client. Internal dialling worked perfectly to our Lync
users but the users could not dial out via our Cisco CME server.
Our configuration is a Lync Enterprise Pool with 2 servers, DB cluster and a single Lync edge server, the mediation servers are installed on our enterprise pool servers as a single server and the CME is looking at the mediation servers directly and has nothing
pointing at the edge server.
This issue affected internal users (as the external users were all kicked out due to the server being down), the strange thing was that you got errors in snooper about the server being unavailable when you dialled out, no idea why it did this. Even stranger
was that the call itself was sent to the CME as a debug SIP showed traffic being attempted between the user and the number which confused me even more as my mobile actually rang for a single ring as well.
Has anybody got any ideas as to why the Edge server would do this to internal users?
Thanks
JamesLync checks the bandwith policy against the Edge server. As the Edge is not responding Lync is unable to check the policy and the call fails.
For the time being you may want to remove the Edge from the topology, then Lync checks against the Front End server.
I hope you do understand, that this is not a great solution, but a drastic workaround suggestion to a hidden product defect! In hidden product defect I mean Microsoft Lync document team is cynically silent and trying to cover the tracks of this product defect.
What I would consider as a straight and honest retroactive action for the Lync document team, to add a big warning section to a) single server edge deployment page + b) the Call Admission Control caveats page on Technet:
"Warning: PRODUCT DEFECT / PRODUCT LIMITATION comes here
If you associate an edge server or pool to a FE pool, and enable Call Admission Control, your single / pool edge will become SINGLE POINT OF FAILURE for your entire enterprise telephony when doing outbound call attempts!
So if outbound calls is important in your company (hell, of course it is!) then deploy at least 2x Edge servers in the same pool before enable CAC!"
But I think that warning message is way too much to ask for, thatswhy is this 2,5 years old topic still open. -
Network Diagram
TX9000 --registered--> CUCM10 --sip trunk--> Telepresence Conductor ----> Telepresence Server VM
CUCM: 192.168.1.11
Telepresence Conductor: 192.168.1.17
TX9000 can call to conference number which match Conference aliases on Telepresence Conductor. But TX9000 video call active only 1 screen. Telepresence Server show log
282 17:09:48.943 APP Info conference "2011" created
283 17:09:49.072 SIP Info Incoming call from 192.168.1.17:53110
284 17:09:49.073 APP Info call 24: new incoming SIP call from "[email protected]"
285 17:09:49.155 APP Info call 24: "1001" now joined conference "2011"
286 17:09:49.172 TIP Warning call 24: not attempting TIP/MUX negotiation (insufficient tokens specified)
287 17:09:49.172 SIP Warning call 24: local failure - attempting to process capabilities while extended video channel is opening
Please suggest me for resolve this issue. Thank youHi,
So you can have an HD resolution - but the requirement here is mutlichannel audio when using multiple screens. So you would need to create a custom quality template with HD and multichannel audio and apply that at the conference template, you can then add the TX9000 with all three screens as HD instead of Full HD.
-Jonathan -
Lync Monitoring Server - Attempted outbound call
Hi,
Im looking at the tables in the monitoring server as a means of monitoring outbound calls where the outgoing agent terminated the call prior to the call being picked up. Is this recorded in the monitoring server? I've checked the ErrorReport which is where
I thought this would land, as well as the sessiondetails table.
Does lync even report such transactions?
Cheers,
Dave.Hi,
You can try to use Call Diagnostic Reports to check the attempted outbound call.
The Call Diagnostic Reports provide summary information and diagnostic data for failed peer-to-peer and conferencing sessions.
More details:
http://social.technet.microsoft.com/Forums/en-US/9d75e7c4-ba5d-4884-8ec8-ee3bd7195821/lync-monitoring-server-attempted-outbound-call?forum=ocsmonitoring
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Unexpected Call Failures during physical network failure
Our layout:
We have a private network for Mediation Server<->PSTN Gateway traffic. (This is a legacy that we are moving away from.)
We have a multi-node FE Pool. The FE Pool has redundant "corp" NICs using Windows Teaming (~failover) that serves clients and our SIP trunk, but a single NIC to the private network for the PSTN Gateway traffic.
The Mediation Seervice is listening on both "corp" and "pstn" NICs.
What happened:
The PSTN network cable for ONE of the mediation servers ("LyncFE1") was cut inadvertently.
The experience:
Approximately 5 minutes after the cable was cut, "LyncFE1" recognized all gateways were offline. (expected)
Users did not lose connectivity to the server, as the "corp" NICs were unaffected. (expected)
Inbound calls routed around the failure, as the PSTN gateways are configured to do. (expected)
Outbound calls for an indeterminate group of people would fail consistently with 503 errors and the logs clearly indicated no routes available. These users were not all hosted on LyncFE1 (per a get-csuserpoolinfo of some of the affected users).
(NOT expected)
I expected that the Mediation Service on "LyncFE1" would recognize that it had no routes, that other servers (let's say "LyncFE2") did have available routes, and that it would then route outbound calls to the other servers via the "corp"
NIC.
Thoughts?Hi,
As the available route is associated with Lync client policy. So If you want to use the other route, you need to associate these users client policy with those route using Lync Server Control Panel or Lync Server Management Shell.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Outbound Call in Cisco Finesse
Anybody please help what all configuration to be done in Finesse to do the outbound calls.
I am using Cisco UCCX 10.0, in which i have both the option for Cisco CAD and Finesse. When i run a outbound campaign its coming as an incoming call in Finesse.
When i answer the call, its showing the error "Unable to communicate with Enterprise Server. Outbound option not available".
Please help how to enable the outbound controls in FinesseThis is ineed the case, from RN (somewhat cryptic :-) ):
Cisco Finesse
Cisco Finesse is the next generation browser-based agent and supervisor desktop for Unified CCX. Finesse is an alternative to Cisco Agent Desktop, Cisco Supervisor Desktop, and Cisco Desktop Administrator. Finesse is available with Enhanced and Premium license packages and provides typical inbound voice contact center functionality. It supports Unified Communications Manager-based silent monitoring and workflow-based recording with MediaSense and Work Force Optimization (WFO).
Chris -
Response Group to PSTN Outbound Caller ID
Hi Guys,
I'm trying to implement an after hours paging service on our lync system using a response group (with overflow set to 0), but the problem I'm currently facing is that the outbound calling number ID is set to the number of the caller, which I'm not legally
allowed to present here in Australia. I could in theory set the number to be private, but that will cause people to screen the calls.
Is there any way to get the RGS application to set the call history in the same way that simultaneous ringing does, even perhaps on some server side-scripting?
This is the SIP info I get in a simultaneous ring, where xxx is the number they called, and yyy being the number they called from.
HISTORY-INFO: <sip:+xxx@frontend;user=phone>;index=1,<sip:+yyy@frontend;user=phone>;index=1.1
Any input would be very appreciated :)I'm curious what happens if you try this:
You'd create a new voice policy, PSTN usage and route. On the route, you mask the caller ID to be your business's primary number.
Then assign that policy to a user account that you've created just for this purpose. That user account should forward it's calls to the number the response group should call.
Then, with the overflow still set to 0, instead of a calling the number, you call that forwarding user account with the caller id mask.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs. -
Bbandtalk sip setup cant make outbound calls
hi i have extracted the sip settings for my bbandtalk service and loaded them into my 3cx server and it has registered fine. However, I cant make outbound calls. It comes up with a message saying 'sorry you cannot make or receive calls on this line'? any ideas? i have seen something about you cant use the bbandtalk sip setup unless you are on your bt broadband line....but if this was so, then how does the bbandtalk softphone client work?
Having similar problems ("Sorry, you cannot make or receive calls ..."). Using FreePBX and Asterisk and could use some help with the settings. Happy to take settings that go into other configuration files, but I'm currently stuck. Here's what I've got thus far (and it doesn't work -- tried several permutations, too):
type=peer
context=incoming_bbt
username=445603449781
authuser=445603449781
secret=B9R0WDKP9KAQMVX
host=62.239.169.148
insecure=very
fromuser=445603449781
fromdomain=62.239.169.148
dtmfmode=rfc2833
Any help would be appreciated. Have other VOIP services working and BT seems to register fine. -
Telepresence server / Conductor cannot go past 5 screen licenses.
I have a IX5000, ex90 and sx80 CUCM 10.5, telepresence server running in VM using the 30 cpu config. It is the latest code. I have 10 license screen on it, For some reason if I try a conference call between the EX90, to SX80 then Conference in the IX5000, the conference fails and I get a license error on the Telepresence server,
I have it set for FULL HD 1080p and content is set same. If I drop to 720p and leave content at 1080p it works. I have not been able to understand why I cannot go past 5 screens. I look at conference and it says I am using 5 screen licenses.
With the configuration I have I should have no issue connecting all three endpoints in a Full HD 1080p conference.
Any one have any ideas? Hope this all makes sense
ex80----calls ix5000- ex80 adds SX90 in and the merge fails when TS is set for 1080pSee page 84 of the admin guide at the following link:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/conductor/admin_guide/TelePresence-Conductor-Admin-Guide-XC3-0.pdf
Field: Maximum screens.
(Available when the Service Preference has a conference bridge type of TelePresence Server and when Allow multiscreen is set to Yes)
For TIP-compliant endpoints dialing into Rendezvous conferences using the TelePresence Conductor's B2BUA, this field specifies the maximum number of screens for which resources are allocated on the conference bridge. The TelePresence Conductor takes the lesser of the Maximum screens value and the number of screens specified by the TIP endpoint and allocates resources accordingly.
For pre-configured endpoints this setting is ignored and the number of screens defined for the pre-configured endpoint are allocated.
For endpoints that are neither TIP-compliant nor pre-configured, this setting is ignored and only a single screen is allocated, unless the endpoint is:
escalated into an ad hoc conference on the TelePresence Conductor
reserved as a host in a Lecture-type conference
using the Cisco VCS's external policy server interface to call into a rendezvous conference
If the endpoint falls into one of the categories listed above, the Maximum screens defines the number of screens for which resources are initially allocated on the conference bridge.
Essentially in an adhoc escalation with 3 screens set on the template, each endpoint will reserve 3 screens worth - or roughly 4 SL per endpoint like the IX5000 is doing at 1080p + 1080p content.
This does not happen in a Rendezvous conference, with 1080p + 1080p content and 3 screens it would take up about 79% of the 10 Screen Licenses and will fit correctly with an IX5000 and two 1 screen endpoints like an SX80
-Jonathan -
FRM-92101: There was a failure in forms server during startup
Hi All,
I installed application server 10g R2 couple of days ago. I am facing FRM-92101: There was a failure in forms server during startup. This could happen due to invalid configuration. Please look into the web server log file for the details.
I am getting this message while calling report on a Form. I have created a button on my main form to call a report and when i press that button i get the above error.
Do i need to do any configuration of application server to call reports.
11/07/05 12:45:44 West Asia Standard Time]::Client Status [ConnId=0, PID=6696]
>> ERROR: Abnormal termination, Error Code: C0000005 ACCESS_VIOLATION
======================= STACK DUMP =======================
Fault address: 6092983C 01:0005883C
Module: L:\oracle\ora10gAS\bin\oranls10.dll
System Information:
Operating System: Windows NT Version 5.0 Build 2195 Service Pack 4
Command line: frmweb server webfile=HTTP-0,0,0,frmtest,192.168.0.13
FORM/BLOCK/FIELD: MAIN_FORM:BLOCK2.ITEM6
Last Trigger: WHEN-BUTTON-PRESSED - (In Progress)
Last Builtin: RUN_REPORT_OBJECT - (In Progress)
Registers:
EAX:0012C048
EBX:00DA2078
ECX:00DACA08
EDX:00000000
ESI:0012C048
EDI:0000000E
CS:EIP:001B:6092983C
SS:ESP:0023:0012BFC8 EBP:0012BFCC
DS:0023 ES:0023 FS:0038 GS:0000
Flags:00210246
------------------- Call Stack Trace ---------------------
Frameptr RetAddr Param#1 Param#2 Param#3 Param#4 Function Name
0x0012bfcc 6648f0fa 0012c048 00000000 00daca08 00da280c _lxscop+4c
------------------- End of Stack Trace -------------------
above is the error dump
Any Idea?[Could you post the block that does the run_report_object?
The server O/S is using what locale?
Also, try running the same report on the command line on the server using rwrun. -
485 Ambiguous - Outbound Calls Only
I'm having some issues with the 485 Ambigous Error, but on Outbound calls only. I've read several blog posts and was able to solve this issue for incoming calls, but have yet to find a solution for outbound calls.
I have two phone numbers: XXX-XXX-5232 and XXX-XXX-3081. All of my users are configured in Lync with +1XXXXXX5232;ext=XXXXX
XXX-XXX-5232 is set to normalize to +1XXXXXX5232;ext=53999 which is a Lync User that is setup with Team Call to ring multiple other extensions. I'm not aware of any AutoAttendant or Response group that is configured with just +1XXXXXX5232.
Again, inbound calling is working just fine and outbound calls work as expected, they just report the 485 Ambiguous for each call.
Thanks in advance for any help you can provide.Yes, the call is routed to the Gateway. In fact, the call completes successfully. Here is the trace. Hopefully it will be readable:
12:17:03.513 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288602 ) ---- Incoming SIP Message from 10.188.0.18:61275 to SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.543 : 10.188.0.19 : NOTICE : INVITE sip:[email protected];user=phone SIP/2.0
FROM: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
TO: <sip:[email protected];user=phone>
CSEQ: 32959 INVITE
CALL-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
CONTACT: <sip:VRRL-SBA.cdol.int:5067;transport=Tls;ms-opaque=86db4f0fd1133b15>
CONTENT-LENGTH: 552
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 885 1 IN IP4 10.188.0.18
s=session
c=IN IP4 10.188.0.18
b=CT:1000
t=0 0
m=audio 53854 RTP/AVP 97 101 13 0 8
c=IN IP4 10.188.0.18
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1
a=rtcp:53855
a=label:Audio
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ssoH69QQto9/wyQyDEbtGezAe4zuH4ulyHNtUfRT|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:Eas2Y5diRZ5HKgxFHpLLTdr8EWMmERj6ZGLjf8LO|2^31
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
[Time: 11:17:03]
12:17:03.573 : 10.188.0.19 : NOTICE : ( sip_stack)(2288604 ) new AcSIPCallAPI created - #276 [Time: 11:17:03]
12:17:03.593 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288605 ) | | new GetNewSIPCall created - #517 [Time: 11:17:03]
12:17:03.603 : 10.188.0.19 : NOTICE : ( lgr_stk_mngr)(2288606 ) Resource StackSession <#276> Allocated [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : ( sip_stack)(2288607 ) TlsTransportObject#57::CheckForConnectionPersistent - Opening persistent connection with proxy: 10.188.0.18:61275 [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288608 ) | |(SIPTU#517)INVITE State:Idle() [Time: 11:17:03]
12:17:03.623 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288609 ) DNSResolver::HandleARecordQuery - Host:VRRL-SBA.cdol.int resolved in external table [Time: 11:17:03]
12:17:03.633 : 10.188.0.19 : NOTICE : ( sip_stack)(2288610 ) (SIPTU#517) HandleResolutionSuccessEV: Domain name VRRL-SBA.cdol.int was successfully resolved to IP: 10.188.0.18 [Time: 11:17:03]
12:17:03.643 : 10.188.0.19 : NOTICE : ( sip_stack)(2288611 ) SIPCall(#517) changes state from Idle to Invited [Time: 11:17:03]
12:17:03.653 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288612 ) | | | #276:SIP_DNS_RESOLVED_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.663 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288613 ) | | | #276:SIP_SETUP_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : ( lgr_call)(2288614 ) (#276) Call Allocated. [Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288615 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 [Time: 11:17:03]
12:17:03.683 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288616 ) <SESSION #276> SendToCall - event: NEW_CALL_EV m_Call#276 [Time: 11:17:03]
12:17:03.693 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288617 ) | | #276:NEW_CALL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.703 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288618 ) | | #276:Call changing states from:IdleState to:NewCallState_IP2Tel [Time:
11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288619 ) ServicesMngr::GetEndPoint PhoneNum = 402XXX0899
[Time: 11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288620 ) GetTrunkGroupId- TrunkGroup:1 found DstNum:402XXX0899 DstPfx:* SrcNum:+1XXXXXX5232 SrcPfx:* SrcIp:abc0012 SrcIpPfx:10.188.0.18 [Time: 11:17:03]
12:17:03.723 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288621 ) QueryOnHookPortStatus (ChannelNum=0), status = 1 Polarity = 0 [Time: 11:17:03]
12:17:03.733 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288622 ) Current trunks status: [Time: 11:17:03]
12:17:03.743 : 10.188.0.19 : NOTICE : ( lgr_num)(2288623 ) PhoneNumber::RemovePrefix - Number change from +1XXXXXX5232 to 1XXXXXX5232 [Time: 11:17:03]
12:17:03.753 : 10.188.0.19 : NOTICE : ( lgr_call)(2288624 ) Call::SetCoderListForCall #276 Found 2 Common Coders For Call [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : ( lgr_call)(2288625 ) <Call #276> Coder g711Ulaw64k20 : 20 [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : ( lgr_call)(2288626 ) <Call #276> Coder g711Alaw64k20 : 20 [Time: 11:17:03]
12:17:03.773 : 10.188.0.19 : NOTICE : ( lgr_profiling)(2288627 ) <Call 276> Profiled<Tel=0,Ip=0>: JBMinDel=10 JBOptF=10 EEarlyM=1 FaxTM=1 IPDS=46 IsFaxU=2 PI2IP=-1 SigIPDF=40 CNGMode=0 DTMFUsed=0 NSEMode=0 PlayRBTone2IP=1
RBUdpPort=0 RTPRD=0 SCE=0 VxxTT=2 Dst2Rdrt=0 DTMFVol=20 ECE=1 ECurDis=0 EDigDel=0 ERevP=0 FHPer=700 InG=32 MWIA=0 MWID=0 VVol=32 ReorderTime=255 DIDWink=0 2StageDial=0 DiscOnBusyT=1 DiscOnBrok=1 DPInd=255 AGC=0 NLP=0 [Time: 11:17:03]
12:17:03.783 : 10.188.0.19 : NOTICE : ( lgr_call)(2288628 ) | | #276GetNextUI:GlobalUI=442334516, mACAddrLsb=3257879 [Time: 11:17:03]
12:17:03.793 : 10.188.0.19 : NOTICE : ( lgr_call)(2288629 ) | | #276GetNextUI:GlobalUI=442334517 [Time: 11:17:03]
12:17:03.803 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288630 ) | #0:NEW_CALL_EV : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288631 ) EndPoint::MediaResourceList::AllocateMediaIpPortsByMediaRealmID Perform NEW allocation of Media ports for RealmIndex(0) port(6220) current allocations
are:(1) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : ( sip_stack)(2288632 ) SIPSDPSession#276 - Changing state from SIP_MEDIA_IDLE to SIP_MEDIA_OFFERED [Time: 11:17:03]
12:17:03.823 : 10.188.0.19 : NOTICE : ( sip_stack)(2288633 ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - CN as Remote 1 [Time: 11:17:03]
12:17:03.833 : 10.188.0.19 : NOTICE : ( sip_stack)(2288634 ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - Force silence suppression on chosen coder, because remote & local support CN [Time: 11:17:03]
12:17:03.843 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288635 ) | |(SIPTU#517)TRYING_REQ State:Invited(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.853 : 10.188.0.19 : NOTICE : ( sip_stack)(2288636 ) New SIPMessage created - #58 [Time: 11:17:03]
12:17:03.863 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288637 ) ---- Outgoing SIP Message to 10.188.0.18:61275 from SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.873 : 10.188.0.19 : NOTICE : SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Length: 0
[Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : ( sip_stack)(2288639 ) Resource SIPMessage deleted - #58 [Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288640 ) SIPStackSession::HandleStackSetupEV - SETUP: SrcPN=0 [Time: 11:17:03]
12:17:03.893 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288641 ) <SESSION #276> SendToCall - event: SETUP_EV m_Call#276 [Time: 11:17:03]
12:17:03.903 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288642 ) | | #276:SETUP (TO:402XXX0899, FROM:+1XXXXXX5232):(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.913 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288643 ) | | #276:Call changing states from:NewCallState_IP2Tel to:InitiatedState_IP2Tel
[Time: 11:17:03]
12:17:03.923 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288644 ) | #0:SETUP_EV : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.933 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288645 ) UpdateChannelParams, Channel 0
[Time: 11:17:03]
12:17:03.943 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288646 ) #0:PSOSBoardInterface::ConfigureFaxModemChannelParams FAXTransportType=3 Modem configuration VxxTransportType=2 not allowed, forced to 3
[Time: 11:17:03]
12:17:03.953 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288647 ) #0:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=3, VxxTranType=3, VoiceVol= 0, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=3, ECNlpMode=0,
DJBufMinDelay=10, DJBufOptFac=10, Result=1) [Time: 11:17:03]
12:17:03.963 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288648 ) Turn ringer ON for channel 0 [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288649 ) | #0:FXO Seize Line [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288650 ) | #0:ALERT_EV (send) : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.983 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288651 ) | | #276:ALERT_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.993 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288652 ) | | #276:Call changing states from:InitiatedState_IP2Tel to:AlertingState_IP2Tel
[Time: 11:17:03]
12:17:04.003 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288653 ) | | | #276:ALERT_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : ( sip_stack)(2288654 ) New SIPMessage created - #93 [Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : ( sip_stack)(2288655 ) SIPSDPSession#276 - Changing state from SIP_MEDIA_OFFERED to SIP_MEDIA_COMPLETED [Time: 11:17:03]
12:17:04.023 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288656 ) DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 11:17:03]
12:17:04.033 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288657 ) DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 11:17:03]
12:17:04.043 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288658 ) <SESSION #276> SendToCall - event: DTMF_CONTROL_EV m_Call#276 [Time: 11:17:03]
12:17:04.053 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288659 ) | | #276:DTMF_CONTROL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time:
11:17:03]
12:17:04.063 : 10.188.0.19 : NOTICE : SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Contact: <sip:[email protected]:5067;transport=tls>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 1
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Type: application/sdp
Content-Length: 254
v=0 -
Outbound call failing with cause code 57
Hi,
our outbound calls to some numbers getting failed with cause code 57 and in ccsip messages i am getting 403 forbidden.
i tried to change the payload type to 97 which was 98 but no success.
the called number is 9-8955900
the calling number is 8062300
can any one help me..
the ccsip messages and ccapi inout debug is..
509022: *Jan 8 14:23:20.513: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255632752,feature_id:53127
509023: *Jan 8 14:23:20.513: //678454/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
509024: *Jan 8 14:23:20.513: //678454/xxxxxxxxxxxx/CCAPI/ccCallSetContext:
Context=0x4AC5D304
509025: *Jan 8 14:23:20.517: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Interface=0x48D4E620, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
509026: *Jan 8 14:23:20.517: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
509027: *Jan 8 14:23:20.537: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:02:15 GMT
Call-ID: [email protected]
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: 2031538944-0000065536-0000027607-3188500672
Session-Expires: 1800
P-Asserted-Identity: "Asif CIPC" <sip:[email protected]>
Remote-Party-ID: "Asif CIPC" <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: 240
v=0
o=CiscoSystemsCCM-SIP 2524413 1 IN IP4 192.168.12.190
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 17706 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
509028: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/cc_api_display_ie_subfield s:
cc_api_call_setup_ind_common:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=3064
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
509029: *Jan 8 14:23:20.553: /
ASICO-DAM#/-1/7916D3000000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x48667600, Call Info(
Calling Number=3064,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T RUE,
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS E), Call Id=678455
509030: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
509031: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
509032: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509033: *Jan 8 14:23:20.553: :cc_get_feature_vsa malloc success
509034: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509035: *Jan 8 14:23:20.553: cc_get_feature_vsa count is 13
509036: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509037: *Jan 8 14:23:20.553: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255634768,feature_id:53128
509038: *Jan 8 14:23:20.553: //678455/7916D3000000/CCAPI/cc_api_call_setup_ind_ common:
Set Up Event Sent;
Call Info(Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passe d, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown))
509039: *Jan 8 14:23:20.557: //678455/7916D3000000/CCAPI/cc_process_call_setup_ ind:
Event=0x48EE6200
509040: *Jan 8 14:23:20.557: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 8955900
509041: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476E35D0
509042: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/cc_process_call_setup_ ind:
>>>>CCAPI handed cid 678455 with tag 1 to app "_ManagedAppProcess_Default"
509043: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
509044: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=20, Params=0x476E4AE0, Progress Indication=NULL(0)
509045: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509046: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509047: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=8955900, Digit Strip=FALSE
509048: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pres entation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Asif CIPC
Account Number=3064, Final Destination Flag=TRUE,
Guid=7916D300-0001-0000-0000-6BD7BE0CA8C0, Outgoing Dial-peer=20
509049: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/cc_api_display_ie_subf ields:
ccCallSetupRequest:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=8062301
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
509050: *Jan 8 14:23:20.569: //678455/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
Interface=0x48667600, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=8062301,(Calling Name=Asif CIPC)(TON=Unknown, NPI= Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20 , Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509051: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509052: *Jan 8 14:23:20.569: :cc_get_feature_vsa malloc success
509053: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509054: *Jan 8 14:23:20.569: cc_get_feature_vsa count is 14
509055: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509056: *Jan 8 14:23:20.569: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255629840,feature_id:53129
509057: *Jan 8 14:23:20.569: //678456/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
509058: *Jan 8 14:23:20.573: //678456/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476E4A90
509059: *Jan 8 14:23:20.573: //678455/7916D3000000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20
509060: *Jan 8 14:23:20.577: //678456/7916D3000000/CCAPI/cc_api_call_proceeding :
Interface=0x48667600, Progress Indication=NULL(0)
509061: *Jan 8 14:23:20.585: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=ye s;privacy=off
From: "Asif CIPC" <sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2031538944-0000065536-0000027607-3188500672
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1389191000
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 274
v=0
o=CiscoSystemsSIP-GW-UserAgent 5380 1731 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 172..XX.XX.XX
t=0 0
m=audio 19502 RTP/AVP 18 101
c=IN IP4 172..XX.XX.XX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
509062: *Jan 8 14:23:20.589: //678455/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
509063: *Jan 8 14:23:20.605: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Call-ID: [email protected]
From: "Asif CIPC"<sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
509064: *Jan 8 14:23:20.677: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Asif CIPC"<sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>;tag=sbc0804k7h28358
CSeq: 101 INVITE
Reason: Q.850;cause=57;text="bearer capability not authorized"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
509065: *Jan 8 14:23:20.677: //678456/7916D3000000/CCAPI/cc_api_call_disconnect ed:
Cause Value=57, Interface=0x48667600, Call Id=678456
509066: *Jan 8 14:23:20.677: //678456/7916D3000000/CCAPI/cc_api_call_disconnect ed:
Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
509067: *Jan 8 14:23:20.681: //678455/7916D3000000/CCAPI/ccCallReleaseResources :
release reserved xcoding resource.
509068: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallSetAAA_Accountin g:
Accounting=0, Call Id=678456
509069: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=57)
509070: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
509071: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48667600, Tag=0x0, Call Id=678456,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
509072: *Jan 8 14:23:20.685: //678456/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509073: *Jan 8 14:23:20.685: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509074: *Jan 8 14:23:20.685: :cc_free_feature_vsa freeing 4AD76408
509075: *Jan 8 14:23:20.685: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509076: *Jan 8 14:23:20.685: vsacount in free is 13
509077: *Jan 8 14:23:20.685: //678455/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=0)
509078: *Jan 8 14:23:20.689: //678455/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
509079: *Jan 8 14:23:20.693: //678455/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>;tag=EA475298-106E
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
509080: *Jan 8 14:23:20.693: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
From: "Asif CIPC" <sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>;tag=sbc0804k7h28358
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
509081: *Jan 8 14:23:20.709: //678454/xxxxxxxxxxxx/CCAPI/ccCallDisconnect:
Cause Value=0, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Ca use=0)
509082: *Jan 8 14:23:20.709: //678454/xxxxxxxxxxxx/CCAPI/ccCallDisconnect:
Cause Value=0, Call Entry(Responsed=TRUE, Cause Value=0)
509083: *Jan 8 14:23:20.709: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>;tag=EA475298-106E
Date: Wed, 08 Jan 2014 14:02:15 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
509084: *Jan 8 14:23:20.713: //678455/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48667600, Tag=0x0, Call Id=678455,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
509085: *Jan 8 14:23:20.717: //678455/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509086: *Jan 8 14:23:20.717: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509087: *Jan 8 14:23:20.717: :cc_free_feature_vsa freeing 4AD77748
509088: *Jan 8 14:23:20.717: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509089: *Jan 8 14:23:20.717: vsacount in free is 12
509090: *Jan 8 14:23:20.721: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48D4E620, Tag=0x0, Call Id=678454,
Call Entry(Disconnect Cause=0, Voice Class Cause Code=0, Retry Count=0)
509091: *Jan 8 14:23:20.721: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509092: *Jan 8 14:23:20.721: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509093: *Jan 8 14:23:20.721: :cc_free_feature_vsa freeing 4AD76F68
509094: *Jan 8 14:23:20.721: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509095: *Jan 8 14:23:20.721: vsacount in free is 11
509096: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivat e:
Interface=0x48D4E620, Interface Type=9, Destination=0.0.0.0, Mode=0x0,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screeni ng=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509097: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509098: *Jan 8 14:23:20.725: :cc_get_feature_vsa malloc success
509099: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509100: *Jan 8 14:23:20.729: cc_get_feature_vsa count is 12
509101: *Jan 8 14:23:20.729: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509102: *Jan 8 14:23:20.729: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255632752,feature_id:53130
509103: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
509104: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/ccCallSetContext:
Context=0x4AC5D1C4
509105: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Interface=0x48D4E620, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
509106: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
509107: *Jan 8 14:23:20.753: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6adee3e42a498
From: "Asif CIPC" <sip:[email protected]>;tag=2524415~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:02:15 GMT
Call-ID: [email protected]
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: 2031538944-0000065536-0000027608-3188500672
Session-Expires: 1800
P-Asserted-Identity: "Asif CIPC" <sip:[email protected]>
Remote-Party-ID: "Asif CIPC" <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: 240
v=0
o=CiscoSystemsCCM-SIP 2524415 1 IN IP4 192.168.12.190
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 17932 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
509108: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/cc_api_display_ie_subfield s:
cc_api_call_setup_ind_common:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=3064
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
509109: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/cc_api_call_setup_ind_comm on:
Interface=0x48667600, Call Info(
Calling Number=3064,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T RUE,
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS E), Call Id=678458
509110: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
509111: *Jan 8 14:23:20.773: //-1/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
509112: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509113: *Jan 8 14:23:20.773: :cc_get_feature_vsa malloc success
509114: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509115: *Jan 8 14:23:20.773: cc_get_feature_vsa count is 13
509116: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509117: *Jan 8 14:23:20.773: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255634768,feature_id:53131
509118: *Jan 8 14:23:20.773: //678458/7916D3000000/CCAPI/cc_api_call_setup_ind_ common:
Set Up Event Sent;
Call Info(Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passe d, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown))
509119: *Jan 8 14:23:20.773: //678458/7916D3000000/CCAPI/cc_process_call_setup_ ind:
Event=0x48EE6200
509120: *Jan 8 14:23:20.777: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 8955900
509121: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476D5190
509122: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/cc_process_call_setup_ ind:
>>>>CCAPI handed cid 678458 with tag 1 to app "_ManagedAppProcess_Default"
509123: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
509124: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=20, Params=0x476DCE60, Progress Indication=NULL(0)
509125: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509126: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509127: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=8955900, Digit Strip=FALSE
509128: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pres entation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Asif CIPC
Account Number=3064, Final Destination Flag=TRUE,
Guid=7916D300-0001-0000-0000-6BD8BE0CA8C0, Outgoing Dial-peer=20
509129: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/cc_api_display_ie_subf ields:
ccCallSetupRequest:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=8062301
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
509130: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
Interface=0x48667600, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=8062301,(Calling Name=Asif CIPC)(TON=Unknown, NPI= Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20 , Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509131: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509132: *Jan 8 14:23:20.785: :cc_get_feature_vsa malloc success
509133: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509134: *Jan 8 14:23:20.785: cc_get_feature_vsa count is 14
509135: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509136: *Jan 8 14:23:20.785: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255629840,feature_id:53132
509137: *Jan 8 14:23:20.789: //678459/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
509138: *Jan 8 14:23:20.789: //678459/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476DCE10
509139: *Jan 8 14:23:20.789: //678458/7916D3000000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20
509140: *Jan 8 14:23:20.793: //678459/7916D3000000/CCAPI/cc_api_call_proceeding :
Interface=0x48667600, Progress Indication=NULL(0)
509141: *Jan 8 14:23:20.801: //678459/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CE197B
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=ye s;privacy=off
From: "Asif CIPC" <sip:[email protected]>;tag=EA475304-26C5
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2031538944-0000065536-0000027608-3188500672
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1389191000
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 274
v=0
o=CiscoSystemsSIP-GW-UserAgent 9218 9584 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 172..XX.XX.XX
t=0 0
m=audio 16868 RTP/AVP 18 101
c=IN IP4 172..XX.XX.XX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
509142: *Jan 8 14:23:20.805: //678458/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
509163: *Jan 8 14:23:20.945: //678458/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509164: *Jan 8 14:23:20.945: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509165: *Jan 8 14:23:20.945: :cc_free_feature_vsa freeing 4AD77748
509166: *Jan 8 14:23:20.945: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509167: *Jan 8 14:23:20.945: vsacount in free is 12
509168: *Jan 8 14:23:32.517: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172..XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKhs8ak5845ch42p7cff35k1ap3T02677
Call-ID: isbchh12748fcsk155w58p151kks36fww24s@SoftX3000
From: <sip:172..XX.XX.XX:5060>;tag=sbc0806pa8fp7w7
To: <sip:172..XX.XX.XX>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
509169: *Jan 8 14:23:32.525: //678460/495AB05FB187/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKhs8ak5845ch42p7cff35k1ap3T02677
From: <sip:172..XX.XX.XX:5060>;tag=sbc0806pa8fp7w7
To: <sip:172..XX.XX.XX>;tag=EA4780CC-582
Date: Wed, 08 Jan 2014 14:23:32 GMT
Call-ID: isbchh12748fcsk155w58p151kks36fww24s@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 170
v=0
o=CiscoSystemsSIP-GW-UserAgent 8937 2437 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.33.5
u all
and the config is
voice service voip
ip address trusted list
ipv4 172.XX.XX.XX 255.255.255.255
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service h225-notify cid-update
redirect ip2ip
h323
session transport udp
h245 tunnel disable
sip
session transport tcp
rel1xx disable
registrar server expires max 3600 min 3500
transport switch udp tcp
redirect contact order best-match
asymmetric payload full
g729 annexb-all
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
and the dial-peer through the calls will go out is
dial-peer voice 20 voip
description ***TO-OUT***
translation-profile outgoing OUT-SIP
destination-pattern .T
progress_ind progress enable 8
rtp payload-type cisco-codec-fax-ack 112
rtp payload-type nte 97
session protocol sipv2
session target ipv4:10.205.20.50:5060
session transport udp
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vadHi Nadeem,
our setup is like
vpn link
call manager ------head office(router)----------------------------------brach router(gateway)------------------ip phone(branch)
The call flow is
sip
IP Phone---->brach GW-------------->call manager------------------>branch GW------------------->ITSP
there is no subscriber at branch side, so the outbound call should travel to call manager at head office and then get exit from branch gateway to ITSP through sip line.
Maybe you are looking for
-
Because I have limited internet package on my att iphone 5, I update apps on my computer thru ITunes and then sync them to my phone. The downloads seems to go thru and then install. But when I turn on my phone, the affected apps continue to say the
-
My old computer crashed and I lost my Ipod, how do I download the music to a new on?
So about 2 years ago I got the Ipod touch, and a while after my computer crashed. I never put the music onto a new computer, because the Ipod allowed me to download music there and then. But now I have lost my Ipod-touch and have a buch of songs sti
-
HI experts, Currently we have the following scenario: There are some raw materials which are procured externally, and these materials are maintained with price control S Well, the problem is that the standard cost for the material consists of a few m
-
Connecting not only to my mac but what about my pc laptop?
connecting works fine but when I try to hook it up to my pc (xp professional) it sees the network but says it cannot connect. Do you have any ideas? gr David
-
I am hoping to transfer data (file size between 70kB~1MB) between two remote locations. I played around with "MB Ethernet Example Master" from ModBus toolkit and was able to successfully send/receive boolean and numeric values. Here are 2 questions: