SIP Routing from CS1000E/T to external Telco SIP Server

[Egghead2]

Hi All,

A customer of mine wants to send all outgoing calls to Telco SIP server and not E1, so I am trying to setup for them.
I am fairly new to the SIP game, but think I have mastered the basics in SIP call routing from CS, SS and NRS.
I have setup all DCH's, routes, trunks, codecs, domains etc. The problem I am getting is that I cannot get my call to route to external SIP server. I have traced DCH and call is being sent to SS/NRS. At NRS I have setup a Gateway Endpoint to external parties SIP server. I have also setup routing to point to this Endpoint.

A trace of the SIPtrace on SS is as follows;

NB : number to go external is mobile 0407665xxx (Australian)
NBB: as this site is over 1000km from me I have setup an indialable ACD 1666 to test. This ACD is NCFW to mobile 00407665xxx.
NBBB: 192.168.1.201 has NAT at firewall to allow it to go external.

oam> 07/10/2009 16:37:24 LOG0006 SIPNPM: SIPCallTrace: 7/10/9 16:37:24 Send chid
:40 ip:192.168.1.204:5060 SIP method INVITE(0)

07/10/2009 16:37:24 LOG0006 SIPNPM: -> INVITE sip:0407665xxx;phone-context=Priv
ateUnknown@DFX;transport=tcp;user=phone SIP/2.0
07/10/2009 16:37:24 LOG0006 SIPNPM: -> From: <sip:407665xxx;phone-context=+61@DF
X;user=phone>;tag=c901a8c0-13c4-4accc3c4-1e898512-7b51
07/10/2009 16:37:24 LOG0006 SIPNPM: -> To: <sip:0407665xxx;phone-context=Privat
eUnknown@DFX;user=phone>
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Call-ID: 253db810-c901a8c0-13c4-4accc3c4-
1e898512-45db@DFX
07/10/2009 16:37:24 LOG0006 SIPNPM: -> CSeq: 1 INVITE
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Via: SIP/2.0/TCP 192.168.1.201:5060;branc
h=z9hG4bK-4accc3c4-1e898512-5fb6
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Max-Forwards: 70
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Supported: 100rel,sipvc,replaces,timer
07/10/2009 16:37:24 LOG0006 SIPNPM: -> User-Agent: Nortel CS1000 SIP GW release_
4.5 version_sse-4.50.88
07/10/2009 16:37:24 LOG0006 SIPNPM: -> P-Asserted-Identity: <sip:407665xxx;phone
-context=+61@DFX;user=phone>
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Privacy: none
07/10/2009 16:37:24 LOG0006 SIPNPM: -> x-nt-e164-clid: +61407665xxx@DFX;user=pho
ne
07/10/2009 16:37:24 LOG0006 SIPNPM: -> History-Info: <sip:1666;phone-context=Pri
vateUnknown@DFX;user=phone?reason=%20sip%3bcause%3d302%3bte
07/10/2009 16:37:24 LOG0006 SIPNPM: ->> xt%3d%20%22Moved%20Temporarily%22>;index
=1,<sip:60407665xxx;phone-context=PrivateUnknown@DFX;user=ph
07/10/2009 16:37:24 LOG0006 SIPNPM: ->> one>;index=2
07/10/2009 16:37:24 LOG0006 SIPNPM: -> x-nt-corr-id: 0000001e102508070a@0001af13
d248-f087c208
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Contact: <sip:407665xxx;phone-context=+61
@DFX:5060;maddr=192.168.1.201;transport=tcp;user=phone>
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTI
FY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Type: multipart/mixed ;boundary=u
nique-boundary-1
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Length: 682
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: -> --unique-boundary-1
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Type: application/SDP
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: -> v=0
07/10/2009 16:37:24 LOG0006 SIPNPM: -> o=- 32 1 IN IP4 192.168.1.201
07/10/2009 16:37:24 LOG0006 SIPNPM: -> s=-
07/10/2009 16:37:24 LOG0006 SIPNPM: -> t=0 0
07/10/2009 16:37:24 LOG0006 SIPNPM: -> m=audio 5202 RTP/AVP 8 0 18
07/10/2009 16:37:24 LOG0006 SIPNPM: -> c=IN IP4 192.168.1.214
07/10/2009 16:37:24 LOG0006 SIPNPM: -> a=fmtp:18 annexb=no
07/10/2009 16:37:24 LOG0006 SIPNPM: -> a=ptime:20
07/10/2009 16:37:24 LOG0006 SIPNPM: -> a=sendrecv
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: -> --unique-boundary-1
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Type: application/x-nt-mcdn-frag-
hex ;version=sse-4.50.88 ;base=x2611
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Disposition: signal ;handling=opt
ional
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 0500a801
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 0107130081900000a2
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 09090f00e9a08300010028
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 1315070011fa0f00a10d02010102020100cc04000
00702
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 200c0f0002008800040001060606
07/10/2009 16:37:24 LOG0006 SIPNPM: -> --unique-boundary-1
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Type: application/x-nt-epid-frag-
hex ;version=sse-4.50.88 ;base=x2611
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Disposition: signal ;handling=opt
ional
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 011201
07/10/2009 16:37:24 LOG0006 SIPNPM: -> 00:02:b3:f7:82:a6
07/10/2009 16:37:24 LOG0006 SIPNPM: -> --unique-boundary-1--
07/10/2009 16:37:24 LOG0006 SIPNPM: SIPCallTrace: 7/10/9 16:37:24 Recv chid:40 i
p:192.168.1.201:5060 SIP response 404

07/10/2009 16:37:24 LOG0006 SIPNPM: -> SIP/2.0 404 Not Found
07/10/2009 16:37:24 LOG0006 SIPNPM: -> From: <sip:407665xxx;phone-context=+61@DF
X;user=phone>;tag=c901a8c0-13c4-4accc3c4-1e898512-7b51
07/10/2009 16:37:24 LOG0006 SIPNPM: -> To: <sip:0407665xxx;phone-context=Privat
eUnknown@DFX;user=phone>;tag=52058
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Call-ID: 253db810-c901a8c0-13c4-4accc3c4-
1e898512-45db@DFX
07/10/2009 16:37:24 LOG0006 SIPNPM: -> CSeq: 1 INVITE
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Via: SIP/2.0/TCP 192.168.1.201:5060;recei
ved=192.168.1.201;branch=z9hG4bK-4accc3c4-1e898512-5fb6
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Contact: <sip:407665xxx;phone-context=+61
@DFX:5060;maddr=192.168.1.201;transport=tcp;user=phone>
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Length: 0
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: SIPCallTrace: 7/10/9 16:37:24 Send chid:40 i
p:192.168.1.204:5060 SIP method ACK(1)

07/10/2009 16:37:24 LOG0006 SIPNPM: -> ACK sip:0407665xxx;phone-context=Private
Unknown@DFX;transport=tcp;user=phone SIP/2.0
07/10/2009 16:37:24 LOG0006 SIPNPM: -> From: <sip:407665xxx;phone-context=+61@DF
X;user=phone>;tag=c901a8c0-13c4-4accc3c4-1e898512-7b51
07/10/2009 16:37:24 LOG0006 SIPNPM: -> To: <sip:0407665xxx;phone-context=Privat
eUnknown@DFX;user=phone>;tag=52058
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Call-ID: 253db810-c901a8c0-13c4-4accc3c4-
1e898512-45db@DFX
07/10/2009 16:37:24 LOG0006 SIPNPM: -> CSeq: 1 ACK
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Via: SIP/2.0/TCP 192.168.1.201:5060;branc
h=z9hG4bK-4accc3c4-1e898512-5fb6
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Max-Forwards: 70
07/10/2009 16:37:24 LOG0006 SIPNPM: -> User-Agent: Nortel CS1000 SIP GW release_
4.5 version_sse-4.50.88
07/10/2009 16:37:24 LOG0006 SIPNPM: -> x-nt-corr-id: 0000001e102508070a@0001af13
d248-f087c208
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Contact: <sip:407665xxx;phone-context=+61
@DFX:5060;maddr=192.168.1.201;transport=tcp;user=phone>
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTI
FY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
07/10/2009 16:37:24 LOG0006 SIPNPM: -> Content-Length: 0
07/10/2009 16:37:24 LOG0006 SIPNPM: ->
07/10/2009 16:37:24 LOG0006 SIPNPM: ->

oam>

As you can see I am getting SIP Response 404.

I look forward to any responses that may help me fix this issue or even point me in the right direction.

 

[gwebster]

Go to

http://www.nortel.com/support

and then do a search on "sip trunk". You may have to go through the results but will eventually find SIP configuration guides for various SIP trunk vendors.

 

[dgrizzard]

I have some experience using Verizon's SIP trunking service, and as far as I know you must have a separate signaling server dedicated to the SIP trunking. I don't think you can just setup a new gw endpoint in NRS that points to the service provider's SIP server.

404 Not Found: Sent from the proxy when either the intended party does not exist (wrong number dialed) or the user is not registered on that proxy.

How are you handling the authentication, sip domain, sip proxy, etc on the NRS to the SIP server?

 

[gwebster]

Generally there are patches required in order to get the correct SIP signalling to the trunking service. If you are using SIP trunking between sites (which uses MCDN buried in the messaging for feature access), then this means you need a separate SIP gateway which is patched for the external trunking service.

 

[Egghead2]

Thanks for the reply guys,

After a bit more research on Nortel site, I agree that I require a separate SS to act as SIP Gateway.
The system I am looking at is a CS1000E/T, and has 2 SS on 1000E, leader and follower, 2 SS on 1000T, leader and follower and also another SS used for NRS only.
I suppose my question is can I use one of the SS in 1000T to act as SIP GW. And if so what patches do I need to load on SS if not already there. Software release on SS is 4.50.88. CS Rel is 4.50W.

 

[dgrizzard]

You could use the SS that is only hosting the NRS, but in the SS configuration do not point it to any H323 gateways, and use the SIP fields for your SIP provider information.

Also, I'm not completely sure as I have 5.5 and I haven't seen anything about 4.5 supporting SIP trunking in a provider type situation. Below is a link to a Nortel PDF on the subject for 5.5 and XO communications as the provider. It does give the patches necessary, but again I've only seen information out there about 5.5 or higher.

SIP Trunking 5.5 XO Communications

 

[Egghead2]

Hi Guys,

After much head banging I can confirm the following (verified by Nortel);

(1) Software release 4.5 does not support the SIP gateway to Telco option.
(2) On later releases a separate SS is required for gateway as this SS requires 3 patches to be loaded in addition to DEPLIST to remove MCDN protocol from signalling. Therefore you cannot use this SS to talk MCDN to any other Nortel device.