Vitual Trunking From SRG > CS1000 E - SIP & H323

[sl1input]

Hi

Im having trouble passing calls between SRG and CS1000 E.

I have so far set up:

CS1000 with SIP trunks registered to NRS.
SRG50 with H323 trunks also registered to NRS.
Configured as a CDP network.
CDP entries in NRS pointing to relevant system.
Testing SRG set in local mode.
Payload 20ms on both systems.
PNI, VPNI, NODE and Zone all same both sides.

I am only able to get the site to register as a H323 endpoint, however the SRG keycode support 24 H323 or SIP trunks.

I am unsure if the NRS supports a H323 to SIP call?

Below is my Sig Server Shell output on the failed calls:

SRG to CS1000 Call (H323 from SRG)

03/11/2006 20:23:03 LOG0006 GKNPM: gkTrace: recv ARQ from 192.168.123.21 calling called 2000
03/11/2006 20:23:03 LOG0005 GKNPM: gkNpmRasError: ARQ, requestDenied, 192.168.123.21:36151, dst 2000(4)
03/11/2006 20:23:03 LOG0006 GKNPM: gkTrace: sent ARJ to 192.168.123.21 calling called 2000, No default route found

CS1000 to SRG Call (SIP From CS1000)


oam> 03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method INVITE(0) chid: 50 Called num: 6421 Far End Signaling IP: 192.168.123.3:5060 Transport:TCP CSeq: 1 INVITE
From: <sip:2000;phone-context=cdp@TEST.COM;user=phone>
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:6421;phone-context=cdp@TEST.COM;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
Media Info: 192.168.123.123 Codecs: G711 A-Law(8) G711 U-Law(0) G729(18) Dynamic(101) Dynamic(111) Payload: 20 ms Media State: SIPNPM_MEDIA_SENDRECV

03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace: This is Incoming Message
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
Message: Incoming response 404 Not Found chid: 50 Called num: 6421 Far End Signaling IP: 192.168.123.3:5060 Transport:TCP CSeq: 1 INVITE
From: <sip:2000;phone-context=cdp@TEST.COM;user=phone>
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:6421;phone-context=cdp@TEST.COM;user=phone>

03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method ACK(1) chid: 50 Called num: 6421 Far End Signaling IP: 192.168.123.3:5060 Transport:TCP CSeq: 1 ACK
From: <sip:2000;phone-context=cdp@TEST.COM;user=phone>
03/11/2006 20:22:15 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:6421;phone-context=cdp@TEST.COM;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31



Thanks

 

[allenmac]

The NRS will support both H.323 and SIP.

I believe your SRG must be configured using SIP in order to pass calls, because your CS1000 does not have any H323 trunks.

 

[Triton101]

Use the NRS call test feature to make sure you dialing plan is not the issue.

 

[sl1input]

When one endpoint is H323 and one SIP the routing test fails both ways.

I have however made the SRG a static SIP endpoint using UDP.
Now if i try SIP routing test it works. However if i call from SRG is see "resource unavail" on IP set. If i call from CS i see "no route to destination" on the DCH monitor.

I don't think the endpoint is properly registered. In the SRG there is no where to assign a SIP Alias name as there is only Alias name in the H323 configuration.

Also where would i tell the SRG to use SIP trunks instead of H323?

 

[edkender]

If you are using sip you need to have a static sip endpoint setup which you do. You also need to make sure your domain name is in there correctly. You also need to make sure you are not trying to route those calls out h323.

 

[sl1input]

Yes i have also set my Domain Name in SRG the same as the Service Domain in the NRS. (i.e. ABC.COM)

I have my VOIP trunks in Bloc:A, a destination code of 2 pointing at route 2, which is using Bloc:A as private.


I have no config in the H323 settings, just the config in SIP settings.

But i am still not clear where i can set the trunks as SIP instead of H323.

 

[edkender]

You do not set it. In the h323 settings if you leave the field gatekeeper digits blank it will route the call sip all the time. If you put something in there it will try to route that h323. Since you only have sip on the cs1000 you need to route it sip.