[cisco-voip] Inbound SIP Calls routing over QSIG link

[Bob Zanett (US)]

I would add MTP resources to the MRGLs on both your SIP  and QSIQ trunks.   This allows CUCM to allocate them as needed and when needed.  You will then want to look into your debugs to validate use.

It is probably a good idea to look at your debugs to validate exactly what is occurring.   You can start at your QSIG gateway - is your call making it this far?   If it is, you should see what is occurring.   Secondly, RTMT on CUCM will allow you to capture what is happening on the QSIG gateway and possibly why the call is failing.   You could also look at the SIP messages on the SIP side in CUCM but my guess is that the error will be rather vague.

Cheers,

Bob
Solutions Architect
Customer Interactive Solutions
Dimension Data
www.dimensiondata.com

From: Nick [mailto:csvoip at googlemail.com]
Sent: Tuesday, February 15, 2011 4:13 PM
To: Bob Zanett (US)
Cc: ROZA, Ariel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Inbound SIP Calls routing over QSIG link

Hi Bob

Ok, I have IOS transcoder and IOS MTP resources setup, where should I apply these, to the MGCP gateway.

Regards

Nick
On 15 February 2011 21:51, Bob Zanett (US) <bob.zanett at us.didata.com<mailto:bob.zanett at us.didata.com>> wrote:
Going between SIP and MGCP - you will most likely require an MTP resource.  A transcoder will perform that but you may not require a full transcode resource.

DTMF:   Usually DTMF has to be converted from RFC2833 (typically used) to MGCP.  There is a command you can place on the MGCP gateway to allow it to understand RFC 2833 DTMF - search on Cisco for MGCP and RFC2833.  However, you would need to validate that all potential endpoints within your Cisco environment can also handle it - otherwise MTP again.

Codec: I would assume your codec is remaining the same either G711 or G729.  If codec changes, then yes you would require a transcode resource.

Bob

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net>] On Behalf Of ROZA, Ariel
Sent: Tuesday, February 15, 2011 3:19 PM
To: Nick; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Inbound SIP Calls routing over QSIG link

Nick,

A Resources Unavailable message sounds like you need to enable the use of transcoding for the call.

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net>] On Behalf Of Nick
Sent: martes, 15 de febrero de 2011 01:24 p.m.
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] Inbound SIP Calls routing over QSIG link

Hi All

I'm trying to route inbound calls from SIP trunks through CUCM to another PBX connected with QSIG, I have CUCM 8.0.3, with QSIG links to a IPC dealerboard system and also a Nortel CS1000 PBX, these are set up as MGCP in CUCM.

I also have SIP trunks for PSTN providing both inbound and outbound calling. The issue I have is when I route inbound calls from the SIP trunks to CUCM and across the QSIG links to the PBX's, the call will not complete and I get Resources Unavailable, Unspecified on the debug on the gateway, I have also taken a SIP trace and get a 503 Service Unavailable from CUCM to the SIP trunks for the same call.

Has anyone successfully done this before and can I route SIP to Qsig or would I need a CUBE even though its going through CUCM.

I have attached the debug and SIP trace.

Any help would be greatly appreciated as this is rather urgent now.

Regards

Nick
________________________________
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