[cisco-voip] SIP Trunk question

[Bob Zanett (US)]

Eric and Jim -
Your (Eric) situation sounds like MTP resources may be running out.   When CUCM is in the loop - my guess you are using software MTPs to handle DTMF, etc negotiations.   By looking in your CUCM trace files will indicate if this is the case.
Two key reasons SIP has problems:

1.       EO versus DO - early offer versus delayed offer     EO requires MTP on CUCM at this time and EO is many times what telco's and some vendors require.

2.       DTMF -  the "norm" is leaning to RFC2833 - though some companies still use other methods.   However, if your applications do not support RFC2833 you will require MTPs.  The phone loads, Unity(as I recall), etc do support it.   In addition, if you are running MGCP gateways and have SIP phones, there are commands to allow MGCP to speak RFC2833.

As a side note:
If you are hitting any issues with transfers, conferencing, etc... simply add router-based software MTPs on the CUBE or ingress gateway and it will resolve most if not all of these issues.

Nick covers some items below.   A couple of others I have seen cause issues:

1.       Make sure that the CUCM Group you have assigned via the device pool on the SIP trunks matches the Subscribers that you are pointing to with your dial peers.   That is if your dial peers point to Subs 1, 2 and 4.  Set up your group for 1,2 and 4.

2.       Transcoding and DTMF - validate that if you are using MTPs that they can handle what you need them to.   Different MTP sources can handle only specific items - see the SRND, etc for more detail.

3.       Nick mentions it below but do make sure your pTime is the same (packetization time).  Early on when integrating CUCM 6.X and Microsoft UM this was an issue depending on how MTPs were used.

Lastly, if you want to reach out to me, I am more than willing to jump on a call to chat.

Cheers,
Bob
Solutions Architect
Customer Interactive Solutions
Dimension Data

www.dimensiondata.com

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Matthews
Sent: Monday, September 27, 2010 6:36 PM
To: Eric Brander
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP Trunk question

95% of the time these are dial peer misconfigurations.  Run 'debug voip dialpeer' in your two situations with no other calls on the router, and see if there is a difference.  If you hit dial peer 0, you need to read about incoming dial peers and fix the problem. Another 3% are just a interop issue and you need to reconfigure the DTMF method from end-to-end, or somewhere in the middle.

In very rare situations timing issues prevent DTMF from happening.  Changing H.323 to slow-start to fast-start or vice versa will sometimes fix that, or using an MTP on CUCM.

-nick
On Mon, Sep 27, 2010 at 6:40 PM, Eric Brander <mailinglists at rednarb.com<mailto:mailinglists at rednarb.com>> wrote:
On Mon, Sep 27, 2010 at 2:53 PM, Jim McBurnett <jim at tgasolutions.com<mailto:jim at tgasolutions.com>> wrote:
> So does anyone have a good idea as to why some calls on a SIP trunk will
> work fine with DTMF and a few fail?
>
>
>
> IE-
>
> Call to X - AA options don't work..
>
> Call to Y-work fine..
>
I'm fighting with sometjhing similar right now in a configuration
between Cisco and NobleSystems (dialer platform based on Asterisk). If
the call placed from my Cisco desk phone communicates directly to a
call manager which then in turn passes the data across the SIP trunk
to Noble then everything works fine. Sometimes however the call will
be handled initially by the call manager but then the phone
communicates directly to the Noble SIP server instead of using the
call manager as an intermediary - when this happens it appears not all
commands are accepted, including DTMF. It seems load related - the
more SIP calls in-place, the more likely for this direct connection
problem.

I can't for the life of me find a solution... but not sure if its
related to your issue or not but sure sounds similar.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip



-----------------------------------------
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/20100928/d8d73950/attachment.html>