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

Bob Zanett (US) bob.zanett at us.didata.com
Tue Feb 15 16:51:28 EST 2011


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] On Behalf Of ROZA, Ariel
Sent: Tuesday, February 15, 2011 3:19 PM
To: Nick; 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] On Behalf Of Nick
Sent: martes, 15 de febrero de 2011 01:24 p.m.
To: 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


-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110215/2c6c64a0/attachment.html>


More information about the cisco-voip mailing list