[cisco-voip] SIP Trunk from ISP DTMF issue

Ahmed Abd EL-Rahman Ahmed.Rahman at bmbgroup.com
Fri Jul 16 17:02:00 EDT 2010


Dear Paul,

                Could you please clarify where exactly the ""Accept
Unsolicited Notification" box of the SIP Trunk Profile" as I can't find
this option in the sip trunk page or in the Standard Sip Profile page.

Please help as I'm suffering from the DTMF problem and I'm using CUCM
7.1(3) with a sip trunk to my VGW which communicates with the Provider
through a sip trunk on the VGW.

 

Best Regards

 

Ahmed Abd EL-Rahman

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul van den
IJssel
Sent: Friday, July 16, 2010 11:09 AM
To: cisco-voip at puck.nether.net"
Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue

 

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#wp
1054848
>>
>>
>>
>> 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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