BrianCosta
Systems Engineer
Hi,
I need an assist me in attempting to complete a SIP call to a Cisco Call Manager (CUCM11.5) from an Avaya IPOSE 11.1. I am able to successfully call from any Avaya IP Phone on the IP Office to any extension on the CCM, but I cannot complete any calls from the CCM to the IPOSE. Here is a sample sys monitor output from a failed call, and How I configured the SIP trunk it from my side, Have I overlooked anything?
11:57:29 519692392mS SIP Call Rx: 5
INVITE sip:1846@10.234.11.1:5060 SIP/2.0
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
To: <sip:1846@10.234.11.1>
Date: Tue, 14 Mar 2023 08:59:06 GMT
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 345a186000105000a0006c030921e027;remote=00000000000000000000000000000000
Cisco-Guid: 2017701504-0000065536-0000060454-0358945964
Session-Expires: 1800
P-Asserted-Identity: "jabber Test" <sip:2222@172.20.101.21>
Remote-Party-ID: "jabber Test" <sip:2222@172.20.101.21>;party=calling;screen=yes;privacy=off
Contact: <sip:2222@172.20.101.21:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP6C030921E027"
Max-Forwards: 69
Content-Length: 0
11:57:29 519692392mS CMCallEvt: 0000000000000000 0.1610614097.0 -1 BaseEP: NEW CMEndpoint f52ea120 TOTAL NOW=1 CALL_LIST=0
11:57:29 519692392mS Sip: SIP Line (5): sip_trunk_config_items 50020001, sip_trunk_config_items_2 00000000, voip.flags 80800949
11:57:29 519692392mS Sip: SIPDialog f52e8538 created, dialogs 3 txn_keys 1 video 1 presentation 1 camera 1 unsupp audio 0
11:57:29 519692392mS CMMap: IP::SetCodec pcp[463]b0r0 0 -> f6ac6ba8
11:57:29 519692393mS Sip: SIP Line (5) GetNetworkTopologySource Use Network Topology badly configured
11:57:29 519692393mS Sip: SIP Line (5) GetNetworkTopologySource Use Network Topology badly configured
11:57:29 519692393mS Sip: SipTCPUser 4297 has 1 dialog open (AttachDialogToSipTCPUser)
11:57:29 519692393mS Sip: SIP Line (5): License, Valid 1, Available 15, Consumed 0
11:57:29 519692393mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692394mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692394mS SIP Call Tx: 5
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692394mS SIP Tx: TCP 10.234.11.1:5060 -> 172.20.101.21:47563
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692394mS Sip: SIP Line (5): Incoming SIP Call Failed.
11:57:29 519692395mS Sip: SIP Line (5): Incoming SIP Call Failed.
11:57:29 519692395mS Sip: 0aea0b0160000551 5.1610614097.1 -1 SIPTrunk Endpoint(f52e8538) Present Call, no match (1846) from URI in To header or (1846) from request URI
11:57:29 519692395mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692395mS SIP Call Tx: 5
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692395mS SIP Tx: TCP 10.234.11.1:5060 -> 172.20.101.21:47563
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692546mS SIP Rx: TCP 172.20.101.21:47563 -> 10.234.11.1:5060
ACK sip:1846@10.234.11.1:5060 SIP/2.0
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Date: Tue, 14 Mar 2023 08:59:06 GMT
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
I know very well that the bug is here :
11:57:29 519692395mS Sip: 0aea0b0160000551 5.1610614097.1 -1 SIPTrunk Endpoint(f52e8538) Present Call, no match (1846) from URI in To header or (1846) from request URI
Any thoughts?
I need an assist me in attempting to complete a SIP call to a Cisco Call Manager (CUCM11.5) from an Avaya IPOSE 11.1. I am able to successfully call from any Avaya IP Phone on the IP Office to any extension on the CCM, but I cannot complete any calls from the CCM to the IPOSE. Here is a sample sys monitor output from a failed call, and How I configured the SIP trunk it from my side, Have I overlooked anything?
11:57:29 519692392mS SIP Call Rx: 5
INVITE sip:1846@10.234.11.1:5060 SIP/2.0
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
To: <sip:1846@10.234.11.1>
Date: Tue, 14 Mar 2023 08:59:06 GMT
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 345a186000105000a0006c030921e027;remote=00000000000000000000000000000000
Cisco-Guid: 2017701504-0000065536-0000060454-0358945964
Session-Expires: 1800
P-Asserted-Identity: "jabber Test" <sip:2222@172.20.101.21>
Remote-Party-ID: "jabber Test" <sip:2222@172.20.101.21>;party=calling;screen=yes;privacy=off
Contact: <sip:2222@172.20.101.21:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP6C030921E027"
Max-Forwards: 69
Content-Length: 0
11:57:29 519692392mS CMCallEvt: 0000000000000000 0.1610614097.0 -1 BaseEP: NEW CMEndpoint f52ea120 TOTAL NOW=1 CALL_LIST=0
11:57:29 519692392mS Sip: SIP Line (5): sip_trunk_config_items 50020001, sip_trunk_config_items_2 00000000, voip.flags 80800949
11:57:29 519692392mS Sip: SIPDialog f52e8538 created, dialogs 3 txn_keys 1 video 1 presentation 1 camera 1 unsupp audio 0
11:57:29 519692392mS CMMap: IP::SetCodec pcp[463]b0r0 0 -> f6ac6ba8
11:57:29 519692393mS Sip: SIP Line (5) GetNetworkTopologySource Use Network Topology badly configured
11:57:29 519692393mS Sip: SIP Line (5) GetNetworkTopologySource Use Network Topology badly configured
11:57:29 519692393mS Sip: SipTCPUser 4297 has 1 dialog open (AttachDialogToSipTCPUser)
11:57:29 519692393mS Sip: SIP Line (5): License, Valid 1, Available 15, Consumed 0
11:57:29 519692393mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692394mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692394mS SIP Call Tx: 5
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692394mS SIP Tx: TCP 10.234.11.1:5060 -> 172.20.101.21:47563
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692394mS Sip: SIP Line (5): Incoming SIP Call Failed.
11:57:29 519692395mS Sip: SIP Line (5): Incoming SIP Call Failed.
11:57:29 519692395mS Sip: 0aea0b0160000551 5.1610614097.1 -1 SIPTrunk Endpoint(f52e8538) Present Call, no match (1846) from URI in To header or (1846) from request URI
11:57:29 519692395mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 172.20.101.21:5060 to 172.20.101.21:5060
11:57:29 519692395mS SIP Call Tx: 5
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692395mS SIP Tx: TCP 10.234.11.1:5060 -> 172.20.101.21:47563
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.3.0 build 47
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Content-Length: 0
11:57:29 519692546mS SIP Rx: TCP 172.20.101.21:47563 -> 10.234.11.1:5060
ACK sip:1846@10.234.11.1:5060 SIP/2.0
Via: SIP/2.0/TCP 172.20.101.21:5060;branch=z9hG4bK2a56212d74e917
From: "jabber Test" <sip:2222@172.20.101.21>;tag=27600801~a3e24dc8-8444-4488-bc75-cbe3658de430-35209806
To: <sip:1846@10.234.11.1>;tag=6a85cf84df11c56d
Date: Tue, 14 Mar 2023 08:59:06 GMT
Call-ID: 7843ae80-4101375a-1f0e3f-156514ac@172.20.101.21
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
I know very well that the bug is here :
11:57:29 519692395mS Sip: 0aea0b0160000551 5.1610614097.1 -1 SIPTrunk Endpoint(f52e8538) Present Call, no match (1846) from URI in To header or (1846) from request URI
Any thoughts?