[cisco-voip] SIP Trunk from ISP DTMF issue

Fri Jul 16 04:36:35 EDT 2010

Sorry, nevermind my previous e-mail. I need hardware transconder to get this
to work..

2010/7/16 Paul van den IJssel <pijssel at gmail.com>

> We got it to work! I checked the "Accept Unsolicited Notification" box of
> the SIP Trunk Profile, and now DTMF is working fine on my in- and outbound
> calls.
>
> Now here's the next challange.. SIP Trunk is in a region running G.711. IP
> Phone A and B are using G.711, IP Phone C and D are using G.729. All devices
> have a MRGL containing both MTP (G.711) and CFB (G.711). Setting up a conf
> call to the G.711 phones is working fine. But when I try to setup a conf
> call to phone A and B, or A and C. The G.729 call gets terminated. It should
> use the CFB and MTP just as it whould with a normal H.323 or MGCP gateway
> connected, right?
>
> Paul
>
>
> 2010/7/15 Nick Matthews <matthnick at gmail.com>
>
> 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
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > cisco-voip mailing list
>> >> > cisco-voip at puck.nether.net
>> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
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>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
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