I have a CS1000 running 5.5 as node 1000 and IP address 192.168.200.38. My NRS is 192.168.200.37. My system xlite client is 192.168.200.100 and the client can receive calls, but it can't make calls out. I have a Cisco CUCM system sip connected to the CS1000 and it can make inbound and outbound calls on that sip route, but I'm not sure where my sip route would be for the xlite client.
I have put a copy of the xlite debug later on in this email so you can see what's happening. Any hints is appreciated. I'm thinking something with a SIP route in the cs1000 to the xlite client, but i'm not sure how/why i'd need to do that.
PreReqs:
1. Nortel SIP Access Ports
2. A working SIP Route in the CS1000
3. A working RLB Entry (RLI) pointing to the SIP Route
4. A Working Nortel/COTS Signaling Server & NRS
5. Your CS1000 configured and recognized as a SIP Endpoint in NRS with correct routing entries relevant to your CS1000
6. Counterpath XLite SIP Softphone installed on a PC that has access to the Nortel CLAN
7. It is VERY important that your Service Domain in the NRS has a DNS entry associated (A & PTR records) that point to the hosting NRS. In this example, sip.company.com needs to resolve to the CLAN IP address of the NRS. Do NOT continue until this step is done.
[09-12-21]16:38:28.642 | Debug | RESIP:TRANSPORT | "Transmitting to [ V4 192.168.100.37:5060 UDP target domain=pbxlab.org received on: Transport: [ V4 0.0.0.0:19868 UDP target domain=unspecified connectionId=0 ] connectionId=0 ] tlsDomain= via [ V4 192.168.100.100:19868 UDP target domain=unspecified connectionId=0 ]
ACK sip:2710;phone-context=interop.rtp@pbxlab.org:5060;maddr=192.168.100.38;transport=tcp;x-nt-redirect=redirect-server SIP/2.0
Via: SIP/2.0/UDP 192.168.100.100:19868;branch=z9hG4bK-d8754z-ee39e269ac33305a-1---d8754z-;rport
To: ""2710"" <sip:2710@pbxlab.org>;tag=46183
From: ""drlucas-5555""<sip:drlucas@pbxlab.org>;tag=357f6f18
Call-ID: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.
CSeq: 2 ACK
Content-Length: 0
" |
[09-12-21]16:38:28.642 | Debug | RESIP:TRANSPORT | "Adding message to tx buffer to: [ V4 192.168.100.37:5060 UDP target domain=pbxlab.org received on: Transport: [ V4 0.0.0.0:19868 UDP target domain=unspecified connectionId=0 ] connectionId=0 ]" |
[09-12-21]16:38:28.642 | Debug | RESIP
NS | "Whitelisting pbxlab.org(1): 192.168.100.37" |
[09-12-21]16:38:28.642 | Debug | RESIPNS | "updating an existing vip: 192.168.100.37 with 192.168.100.37" |
[09-12-21]16:38:28.642 | Debug | RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=0
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.100.100:19868;branch=z9hG4bK-d8754z-ee39e269ac33305a-1---d8754z-;rport;received=192.168.100.100
Contact: <sip:2710;phone-context=interop.rtp@pbxlab.org:5060;maddr=192.168.100.38;transport=tcp;x-nt-redirect=redirect-server>
To: ""2710"" <sip:2710@pbxlab.org>;tag=46183
From: ""drlucas-5555"" <sip:drlucas@pbxlab.org>;tag=357f6f18
Call-ID: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.
CSeq: 2 INVITE
Content-Length: 0
" |
[09-12-21]16:38:28.643 | Info | RESIPUM | "Got: SipResp: 302 tid=ee39e269ac33305a cseq=INVITE contact=2710;phone-context=interop.rtp@pbxlab.org:5060 / 2 from(wire)" |
[09-12-21]16:38:28.643 | Debug | RESIPUM | "DialogUsageManager:rocessResponse:
SipResp: 302 tid=ee39e269ac33305a cseq=INVITE contact=2710;phone-context=interop.rtp@pbxlab.org:5060 / 2 from(wire)" |
[09-12-21]16:38:28.643 | Debug | CCM | "[EOfferSent] [drlucas@pbxlab.org -> 2710@pbxlab.org]" | sua::CSIPOutgoingCallGroup:nRedirectReceived
[09-12-21]16:38:28.643 | Debug | RESIPUM | "DialogId:ialogId: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.-357f6f18-46183" |
[09-12-21]16:38:28.643 | Info | RESIPUM | "Cannot create a dialog, cseq does not match initial dialog request (illegal mid-dialog fork? see 3261 14.1)." |
[09-12-21]16:38:28.648 | Info | Audio | "Evaluating audio 638 1" | sua::CAudioManager::ServiceAudioOut
I have put a copy of the xlite debug later on in this email so you can see what's happening. Any hints is appreciated. I'm thinking something with a SIP route in the cs1000 to the xlite client, but i'm not sure how/why i'd need to do that.
PreReqs:
1. Nortel SIP Access Ports
2. A working SIP Route in the CS1000
3. A working RLB Entry (RLI) pointing to the SIP Route
4. A Working Nortel/COTS Signaling Server & NRS
5. Your CS1000 configured and recognized as a SIP Endpoint in NRS with correct routing entries relevant to your CS1000
6. Counterpath XLite SIP Softphone installed on a PC that has access to the Nortel CLAN
7. It is VERY important that your Service Domain in the NRS has a DNS entry associated (A & PTR records) that point to the hosting NRS. In this example, sip.company.com needs to resolve to the CLAN IP address of the NRS. Do NOT continue until this step is done.
[09-12-21]16:38:28.642 | Debug | RESIP:TRANSPORT | "Transmitting to [ V4 192.168.100.37:5060 UDP target domain=pbxlab.org received on: Transport: [ V4 0.0.0.0:19868 UDP target domain=unspecified connectionId=0 ] connectionId=0 ] tlsDomain= via [ V4 192.168.100.100:19868 UDP target domain=unspecified connectionId=0 ]
ACK sip:2710;phone-context=interop.rtp@pbxlab.org:5060;maddr=192.168.100.38;transport=tcp;x-nt-redirect=redirect-server SIP/2.0
Via: SIP/2.0/UDP 192.168.100.100:19868;branch=z9hG4bK-d8754z-ee39e269ac33305a-1---d8754z-;rport
To: ""2710"" <sip:2710@pbxlab.org>;tag=46183
From: ""drlucas-5555""<sip:drlucas@pbxlab.org>;tag=357f6f18
Call-ID: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.
CSeq: 2 ACK
Content-Length: 0
" |
[09-12-21]16:38:28.642 | Debug | RESIP:TRANSPORT | "Adding message to tx buffer to: [ V4 192.168.100.37:5060 UDP target domain=pbxlab.org received on: Transport: [ V4 0.0.0.0:19868 UDP target domain=unspecified connectionId=0 ] connectionId=0 ]" |
[09-12-21]16:38:28.642 | Debug | RESIP
NS | "Whitelisting pbxlab.org(1): 192.168.100.37" | [09-12-21]16:38:28.642 | Debug | RESIP
NS | "updating an existing vip: 192.168.100.37 with 192.168.100.37" | [09-12-21]16:38:28.642 | Debug | RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=0
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.100.100:19868;branch=z9hG4bK-d8754z-ee39e269ac33305a-1---d8754z-;rport;received=192.168.100.100
Contact: <sip:2710;phone-context=interop.rtp@pbxlab.org:5060;maddr=192.168.100.38;transport=tcp;x-nt-redirect=redirect-server>
To: ""2710"" <sip:2710@pbxlab.org>;tag=46183
From: ""drlucas-5555"" <sip:drlucas@pbxlab.org>;tag=357f6f18
Call-ID: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.
CSeq: 2 INVITE
Content-Length: 0
" |
[09-12-21]16:38:28.643 | Info | RESIP
UM | "Got: SipResp: 302 tid=ee39e269ac33305a cseq=INVITE contact=2710;phone-context=interop.rtp@pbxlab.org:5060 / 2 from(wire)" | [09-12-21]16:38:28.643 | Debug | RESIP
UM | "DialogUsageManager:rocessResponse: SipResp: 302 tid=ee39e269ac33305a cseq=INVITE contact=2710;phone-context=interop.rtp@pbxlab.org:5060 / 2 from(wire)" |
[09-12-21]16:38:28.643 | Debug | CCM | "[EOfferSent] [drlucas@pbxlab.org -> 2710@pbxlab.org]" | sua::CSIPOutgoingCallGroup:
nRedirectReceived[09-12-21]16:38:28.643 | Debug | RESIP
UM | "DialogId:ialogId: YjkzNTAyMjY2OGIyM2U1YTRmN2Y5YjgyNjhiMDJhOWY.-357f6f18-46183" | [09-12-21]16:38:28.643 | Info | RESIP
UM | "Cannot create a dialog, cseq does not match initial dialog request (illegal mid-dialog fork? see 3261 14.1)." | [09-12-21]16:38:28.648 | Info | Audio | "Evaluating audio 638 1" | sua::CAudioManager::ServiceAudioOut
