[cisco-voip] SIP Trunk from ISP DTMF issue

[Nick Matthews]

You really have to check the DTMF settings on every hop.  I thought
there was a CUE involved also?

If you're not seeing the 101 PT in the SDP in the 200 OK you need to
check the INVITE to see what PT is being advertised, and if it's being
advertised at all.

SIP INFO and NOTIFY are not generally widely used.  I would work with
getting 2833 to work first.

-nick

On Thu, Jul 15, 2010 at 4:58 AM, Paul van den IJssel <pijssel at gmail.com> wrote:
> Hi all,
>
>
>
> Here is some aditional information from the ISP side. they are using a
> Genband MSX release 4.3m6 and we have the following options for DTMF
> transmission:
>
>
>
> SIP NOTIFY
>
> SIP INFO
>
> RFC2833 (payload type 101)
>
>
>
> When we send DTMF to the CCM with SIP NOTIFY we see a 403 Forbidden comming
> back from the CCM.
>
>
>
> One other thing i noticed is that when we place a inbound call to the CCM
> --> IVR (using RFC2833) the 200OK doesen't contain a media type 101
> (rfc2833), and if i'm correct the CCM was configured for RFC2833 at that
> point. If the SBC doesen't receive the correct media type in the 200OK it
> assumes that there is no support for RFC2833 and it will fallback to inband
> DTMF tones in the audio stream.
>
>
>
> Could this have something to do with the DTMF transmission between the CCM
> and unity express? Is the DTMF transmitted in the siggnaling between the CCM
> and the unity express? Or is it using RTP payload like RFC2833 does?
>
>
>
> Here also a brief description of our setup:
>
>
>
>
>
> [ISDN30/DMS100]--------------------[Cisco AS5850]------------------[SIP
> proxy]--------------------[Genband SBC]-----------------[CCM]
>
>                                ISDN30
> SIP                            SIP                                       SIP
>
>
>
>
>
> Thanks in advance,
>
>
>
> Jan Hazenberg
>
>
>
> 2010/7/15 Nicholas Samios <nsamios at staff.iinet.net.au>
>>
>> What codec are you using ?  i.e. Region settings, etc.
>>
>>
>>
>> >> If you are not using a cube, you might need an enhanced IOS software
>> >> MTP termination point which will allow the capturing of DTMF packets inband
>> >> and process them out of band.  These can run as software MTPs on the gateway
>> >> (cannot use the >>UCM built in >>software MTPs for this purpose).
>>
>>
>>
>> That’s incorrect.  CUCM’s inbuilt MTPs can be used to convert DTMF i.e.
>> take OOB H245 and make it RFC2833, etc.
>>
>>
>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/media.html#wp1054848
>>
>>
>>
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul van den IJssel
>> Sent: Thursday, July 15, 2010 4:21 PM
>> To: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue
>>
>>
>>
>> My ISP is able to set the DTMF to:
>>
>> - SIP INFO;
>>
>> - SIP NOTIFY;
>>
>> - RFC2833 (payload type 101);
>>
>>
>>
>> If my ISP sets it to RFC2833 the CUCM should accept it right?
>>
>>
>>
>> 2010/7/13 Matt Slaga (US) <Matt.Slaga at us.didata.com>
>>
>> If you are not using a cube, you might need an enhanced IOS software MTP
>> termination point which will allow the capturing of DTMF packets inband and
>> process them out of band.  These can run as software MTPs on the gateway
>> (cannot use the UCM built in software MTPs for this purpose).
>>
>> ________________________________________
>> From: cisco-voip-bounces at puck.nether.net
>> [cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Matthews
>> [matthnick at gmail.com]
>> Sent: Tuesday, July 13, 2010 11:38 AM
>> To: Mark Holloway
>> Cc: cisco-voip at puck.nether.net; Paul van den IJssel
>> Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue
>>
>> Also check the ccn subsystem sip on the CUE - make sure you're
>> configured for the DTMF method you're using everywhere else.  Plus use
>> a CUBE as stated.
>>
>> -nick
>>
>> On Tue, Jul 13, 2010 at 11:17 AM, Mark Holloway <mh at markholloway.com>
>> wrote:
>> > The Genband is probably doing the same thing that CUBE would be doing.
>> >  What
>> > are the DTMF requirements from your provider?  Have you confirmed that
>> > DTMF
>> > is being passed to the provider?  A Wireshark capture on the public side
>> > of
>> > the S3 would verify DTMF is being passed.
>> > On Jul 13, 2010, at 7:58 AM, Bill wrote:
>> >
>> > You should not trunk directly to CUCM from your ITSP. You need to go
>> > through
>> > a CUBE router. Terminate the ITSP on the CUBE and then do a SIP trunk to
>> > your CUBE router.
>> >
>> > ________________________________
>> > From: cisco-voip-bounces at puck.nether.net
>> > [mailto:cisco-voip-bounces at puck.nether.net] On
>> > Behalf Of Paul van den IJssel
>> > Sent: Tuesday, July 13, 2010 9:54 AM
>> > To: cisco-voip at puck.nether.net
>> > Subject: [cisco-voip] SIP Trunk from ISP DTMF issue
>> >
>> > Hi all,
>> >
>> > Currently I'm running a acceptance test on a SIP Trunk from our ISP.
>> > Everything is working fine except the DTMF.
>> >
>> > Setup:
>> > [GenBand SBC] ---- [SIP Trunk into CUCM 7.1] ---- [Cisco Unity Express
>> > 7.0]
>> >
>> > The SIP Trunk as well as the CUE are in the same region supporting only
>> > G.711. On the SIP Trunk I've tried all different kind of DTMF
>> > configurations
>> > (No Preference, RFC 2833 and OOB). We've tried to force the GenBand SBC
>> > to
>> > only use RFC 2833 as well as the SIP Trunk. But this didn't work, not
>> > even
>> > after we added a MRGL with MTP's.
>> >
>> > Is there some sort of best practive to implement DTMF over a SIP Trunk?
>> > Is
>> > there a way I can do some debugging on the CUCM/CUE?
>> >
>> > Kind regards,
>> >
>> > Paul van den IJssel
>> > Digacom
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>> >
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>> >
>>
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