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IPO R9.0.1 IVR SIP trunk 2

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Not open for further replies.
cjjb

cjjb

Programmer
Oct 31, 2007
15
US
Hi, I am having a problem with an NISC call capture server trying to dial into the IPO get an outside line for mass call out.
A call coming from the IVR to the IPO is not passing the dialed digits to ARS. The call comes in on SIP line 18 to an incoming call route with the destination set to User "Call Capture". This user is unconditionally forwarded to short code *99. Short code is set as follows.
Code: *99
Feature: Dial
Telephne Number: 8N
Line Group ID: 50:[8]
The call then fails due to digits dialed not coming through.

UNICODE-UTF8
enu

********** Warning: Logging to Screen Started **********

********** SysMonitor v9.0.0.0 build 829 [connected to x.x.x.x ] **********171374377mS CMExtnEvt: v=1011 State, new=Idle old=PortRecoverDelay,0,0,RAS
171374379mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d45b58) received CMSetup
171374381mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d0a040) SetLocalRTPAddress to 10.15.31.79:49152
171374382mS SIP Call Tx: 18
INVITE sip:210@10.15.31.77 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Contact: "HTC" <sip:86189399215@10.15.31.79:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 296

v=0
o=UserA 943483814 3829789000 IN IP4 10.15.31.79
s=Session SDP
c=IN IP4 10.15.31.79
t=0 0
m=audio 49152 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
171374383mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
INVITE sip:210@10.15.31.77 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Contact: "HTC" <sip:86189399215@10.15.31.79:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 296

v=0
o=UserA 943483814 3829789000 IN IP4 10.15.31.79
s=Session SDP
c=IN IP4 10.15.31.79
t=0 0
m=audio 49152 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
171374385mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d45b58) received CMFacility
171374387mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.15.31.79:5060;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6;received=10.15.31.79;rport=5060
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Server: Asterisk PBX 10.3.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:210@10.15.31.77:5060>
Content-Length: 0

171374389mS SIP Call Rx: 18
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.15.31.79:5060;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6;received=10.15.31.79;rport=5060
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Server: Asterisk PBX 10.3.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:210@10.15.31.77:5060>
Content-Length: 0

171374393mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.79:5060;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6;received=10.15.31.79;rport=5060
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Server: Asterisk PBX 10.3.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:210@10.15.31.77:5060>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1564449786 1564449786 IN IP4 10.15.31.77
s=Asterisk PBX 10.3.1
c=IN IP4 10.15.31.77
t=0 0
m=audio 19846 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
171374396mS SIP Call Rx: 18
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.79:5060;branch=z9hG4bK3b74bae86602b0ef0d3c30baa9d973c6;received=10.15.31.79;rport=5060
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 INVITE
Server: Asterisk PBX 10.3.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:210@10.15.31.77:5060>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1564449786 1564449786 IN IP4 10.15.31.77
s=Asterisk PBX 10.3.1
c=IN IP4 10.15.31.77
t=0 0
m=audio 19846 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
171374398mS SIP Call Tx: 18
ACK sip:210@10.15.31.77:5060 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bK4d5080a9c7737a8d95711df8909e8582
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 ACK
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 0

171374398mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
ACK sip:210@10.15.31.77:5060 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bK4d5080a9c7737a8d95711df8909e8582
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546877 ACK
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 0

171374399mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d0a040) SetRfc2833TxPayload: use RFC2833 for dtmf
171374400mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d0a040) SetRemoteRTPAddress to 10.15.31.77:19846
171374409mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d45b58) received CMConnectAck
171378931mS CMExtnEvt: v=10 State, new=PortRecoverDelay old=Connected,0,0,Extn120
171378933mS CMExtnEvt: Extn120: CALL LOST (CMCauseNormal)
171378933mS CMExtnEvt: Extn120: Extn(120) Calling Party Number(120) Type(CMNTypeInternal)
171378933mS CMExtnEvt: Extn120: CMExtnHandler::SetCurrent( id: 5721->0 )
171380934mS CMExtnEvt: Extn120: Recover Timer reason=CMTRWrapUp
171380934mS CMExtnEvt: v=10 State, new=Idle old=PortRecoverDelay,0,0,Extn120
171390924mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
INVITE sip:9391231@10.15.31.79 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>
Contact: <sip:86189399215@10.15.31.77:5060>
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.1
Date: Thu, 27 Mar 2014 19:10:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 513134529 513134529 IN IP4 10.15.31.77
s=Asterisk PBX 10.3.1
c=IN IP4 10.15.31.77
t=0 0
m=audio 15856 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
171390927mS SIP Call Rx: 18
INVITE sip:9391231@10.15.31.79 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>
Contact: <sip:86189399215@10.15.31.77:5060>
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.1
Date: Thu, 27 Mar 2014 19:10:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 513134529 513134529 IN IP4 10.15.31.77
s=Asterisk PBX 10.3.1
c=IN IP4 10.15.31.77
t=0 0
m=audio 15856 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
171390928mS Sip: SIPDialog f4d08998 created, size 2
171390929mS Sip: License, Valid 1, Available 12, Consumed 1
171390931mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.15.31.77:5060 to 10.15.31.77:5060
171390932mS SIP Call Tx: 18
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171390932mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171390933mS Sip: 18.5802.1 -1 SIPTrunk Endpoint(f4d08998) SetRfc2833TxPayload: use RFC2833 for dtmf
171390934mS Sip: 18.5802.1 -1 SIPTrunk Endpoint(f4d08998) SetRemoteRTPAddress to 10.15.31.77:15856
171390939mS CMARS: FindActiveARSByGroupID GroupID=50 - Found
171390939mS CMARS: MakeCallTarget - Called Number: 8
171390941mS Sip: 18.5802.1 443 SIPTrunk Endpoint(f4ce4624) received CMProceeding
171390943mS CMARS: FORM: [8] - Received Number: 8
171390943mS CMARS: FOUND A SHORT CODE - short_code: N; - Tel: N - Called_Party: 8 - Line Group Id: 0
171390943mS CMARS: FindActiveARSByGroupID GroupID=0 - Not Found
171390945mS CMARS: FOUND LINE - Line Id: 203 - using line group id: 0 - Called Number: 8 - Calling Number: 86189399215@10.15.31.77
171390945mS CMARS: SEND Setup TO LINE
171391158mS CMARS: LINE ep Received: CMSetupAck - child->state = CMCSOffering - ARS Call State = CMCSOverlapRecv
171391185mS CMARS: LINE ep Received: CMProceeding - child->state = CMCSOverlapRecv - ARS Call State = CMCSOverlapRecv
171394513mS CMARS: LINE ep Received: CMProgress - child->state = CMCSOverlapRecv - ARS Call State = CMCSOverlapRecv
171394515mS Sip: 18.5802.1 443 SIPTrunk Endpoint(f4ce4624) received CMAlerting
171394515mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.15.31.77:5060 to 10.15.31.77:5060
171394516mS SIP Call Tx: 18
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Contact: <sip:9391231@10.15.31.79:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171394517mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Contact: <sip:9391231@10.15.31.79:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171394518mS CMARS: LINE ep Received: CMAlerting - child->state = CMCSAccept - ARS Call State = CMCSAccept
171394518mS Sip: 18.5802.1 443 SIPTrunk Endpoint(f4ce4624) received CMFacility
171394546mS CMARS: LINE ep Received: CMConnect - child->state = CMCSRinging - ARS Call State = CMCSRinging
171394546mS CMARS: CMARSEndpoint::CallLost(cause=124) - Address: 0.5804.0 443 ARS for [8] - Call State: CMCSRinging
171394550mS Sip: 18.5802.1 443 SIPTrunk Endpoint(f4ce4624) received CMConnect
171394550mS Sip: 18.5802.1 443 SIPTrunk Endpoint(f4d08998) SetLocalRTPAddress to 10.15.31.79:49154
171394550mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.15.31.77:5060 to 10.15.31.77:5060
171394553mS SIP Call Tx: 18
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Contact: <sip:9391231@10.15.31.79:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Type: application/sdp
Content-Length: 202

v=0
o=UserA 1003824245 3770392148 IN IP4 10.15.31.79
s=Session SDP
c=IN IP4 10.15.31.79
t=0 0
m=audio 49154 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
171394553mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK451b748e
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 INVITE
Contact: <sip:9391231@10.15.31.79:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Type: application/sdp
Content-Length: 202

v=0
o=UserA 1003824245 3770392148 IN IP4 10.15.31.79
s=Session SDP
c=IN IP4 10.15.31.79
t=0 0
m=audio 49154 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
171394559mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
ACK sip:9391231@10.15.31.79:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK75695794
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Contact: <sip:86189399215@10.15.31.77:5060>
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.3.1
Content-Length: 0

171394561mS SIP Call Rx: 18
ACK sip:9391231@10.15.31.79:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK75695794
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Contact: <sip:86189399215@10.15.31.77:5060>
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.3.1
Content-Length: 0

171401566mS Sip: 18.5800.0 442 SIPTrunk Endpoint(f4d45b58) received CMFacility
171401571mS Sip: 18.5800.0 -1 SIPTrunk Endpoint(f4d45b58) received CMReleaseComp
171401572mS SIP Call Tx: 18
BYE sip:210@10.15.31.77:5060 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bKb6895b5bf4610da20e562d1d99b0e3e0
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546878 BYE
Contact: "HTC" <sip:86189399215@10.15.31.79:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 0

171401573mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
BYE sip:210@10.15.31.77:5060 SIP/2.0
Via: SIP/2.0/UDP 10.15.31.79:5060;rport;branch=z9hG4bKb6895b5bf4610da20e562d1d99b0e3e0
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546878 BYE
Contact: "HTC" <sip:86189399215@10.15.31.79:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.0.0 build 829
Content-Length: 0

171401573mS Sip: SIPDialog f4d0a040 destroyed, size 1
171401577mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.79:5060;branch=z9hG4bKb6895b5bf4610da20e562d1d99b0e3e0;received=10.15.31.79;rport=5060
From: "HTC" <sip:86189399215@10.15.31.77>;tag=ecf561e1868f38be
To: <sip:210@10.15.31.77>;tag=as5f4157d4
Call-ID: 69b68d12b32dac4ad87da725bd3bf238
CSeq: 1792546878 BYE
Server: Asterisk PBX 10.3.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

171401604mS SIP Rx: UDP 10.15.31.77:5060 -> 10.15.31.79:5060
BYE sip:9391231@10.15.31.79:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK3160d43a
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 10.3.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

171401606mS SIP Call Rx: 18
BYE sip:9391231@10.15.31.79:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK3160d43a
Max-Forwards: 70
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 10.3.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

171401608mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.15.31.77:5060 to 10.15.31.77:5060
171401608mS SIP Call Tx: 18
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK3160d43a
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 103 BYE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171401609mS SIP Tx: UDP 10.15.31.79:5060 -> 10.15.31.77:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.31.77:5060;branch=z9hG4bK3160d43a
From: "HTC" <sip:86189399215@10.15.31.77>;tag=as0947e83c
Call-ID: 31d869d56c02c713350618166b133121@10.15.31.77:5060
CSeq: 103 BYE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:9391231@10.15.31.79>;tag=ee24b563afcf4409
Content-Length: 0

171401613mS Sip: SIPDialog f4d08998 destroyed, size 0

********** Warning: Logging to Screen Stopped **********

Any help would be greatly appreciated.
 
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I have this exact same thing I'm about to setup with NICS. Were you able to figure out a solution?
 
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That was aimed at the OP.....didn't notice the date lol, still true though :)

 
Upvote 0 Downvote
I haven't attempted this yet. I was just looking for some ideas on how to make this work. If the IVR connects to the IPO on a SIP trunk how I can make it dial out over the PRI? How would you setup a short code for that?
 
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You create an Incoming Call route to do it and that matches digits received against existing system shortcodes, it's easy actually:
Incoming line group - SIP trunks
Incoming number - blank
Destination - . (just a dot)

Done :)

 
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