Sorry, nevermind my previous e-mail. I need hardware transconder to get this to work..<br><br>
<div class="gmail_quote">2010/7/16 Paul van den IJssel <span dir="ltr"><<a href="mailto:pijssel@gmail.com">pijssel@gmail.com</a>></span><br>
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<div>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.</div>
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<div>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?</div>
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<div>Paul</div>
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<div class="gmail_quote">2010/7/15 Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></span>
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<div class="h5"><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">You really have to check the DTMF settings on every hop. I thought<br>there was a CUE involved also?<br><br>
If you're not seeing the 101 PT in the SDP in the 200 OK you need to<br>check the INVITE to see what PT is being advertised, and if it's being<br>advertised at all.<br><br>SIP INFO and NOTIFY are not generally widely used. I would work with<br>
getting 2833 to work first.<br><font color="#888888"><br>-nick<br></font>
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<div><br>On Thu, Jul 15, 2010 at 4:58 AM, Paul van den IJssel <<a href="mailto:pijssel@gmail.com" target="_blank">pijssel@gmail.com</a>> wrote:<br>> Hi all,<br>><br>><br>><br>> Here is some aditional information from the ISP side. they are using a<br>
> Genband MSX release 4.3m6 and we have the following options for DTMF<br>> transmission:<br>><br>><br>><br>> SIP NOTIFY<br>><br>> SIP INFO<br>><br>> RFC2833 (payload type 101)<br>><br>><br>
><br>> When we send DTMF to the CCM with SIP NOTIFY we see a 403 Forbidden comming<br>> back from the CCM.<br>><br>><br>><br>> One other thing i noticed is that when we place a inbound call to the CCM<br>
> --> IVR (using RFC2833) the 200OK doesen't contain a media type 101<br>> (rfc2833), and if i'm correct the CCM was configured for RFC2833 at that<br>> point. If the SBC doesen't receive the correct media type in the 200OK it<br>
> assumes that there is no support for RFC2833 and it will fallback to inband<br>> DTMF tones in the audio stream.<br>><br>><br>><br>> Could this have something to do with the DTMF transmission between the CCM<br>
> and unity express? Is the DTMF transmitted in the siggnaling between the CCM<br>> and the unity express? Or is it using RTP payload like RFC2833 does?<br>><br>><br>><br>> Here also a brief description of our setup:<br>
><br>><br>><br>><br>><br>> [ISDN30/DMS100]--------------------[Cisco AS5850]------------------[SIP<br>> proxy]--------------------[Genband SBC]-----------------[CCM]<br>><br>> ISDN30<br>
> SIP SIP SIP<br>><br>><br>><br>><br>><br>> Thanks in advance,<br>><br>><br>><br>> Jan Hazenberg<br>><br>><br>><br>
> 2010/7/15 Nicholas Samios <<a href="mailto:nsamios@staff.iinet.net.au" target="_blank">nsamios@staff.iinet.net.au</a>><br>>><br>>> What codec are you using ? i.e. Region settings, etc.<br>>><br>
>><br>>><br>>> >> If you are not using a cube, you might need an enhanced IOS software<br>>> >> MTP termination point which will allow the capturing of DTMF packets inband<br>>> >> and process them out of band. These can run as software MTPs on the gateway<br>
>> >> (cannot use the >>UCM built in >>software MTPs for this purpose).<br>>><br>>><br>>><br>>> That’s incorrect. CUCM’s inbuilt MTPs can be used to convert DTMF i.e.<br>>> take OOB H245 and make it RFC2833, etc.<br>
>><br>>><br>>> <a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/media.html#wp1054848" target="_blank">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/media.html#wp1054848</a><br>
>><br>>><br>>><br>>> From: <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a><br>>> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Paul van den IJssel<br>
>> Sent: Thursday, July 15, 2010 4:21 PM<br>>> To: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>>> Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue<br>
>><br>>><br>>><br>>> My ISP is able to set the DTMF to:<br>>><br>>> - SIP INFO;<br>>><br>>> - SIP NOTIFY;<br>>><br>>> - RFC2833 (payload type 101);<br>>><br>
>><br>>><br>>> If my ISP sets it to RFC2833 the CUCM should accept it right?<br>>><br>>><br>>><br>>> 2010/7/13 Matt Slaga (US) <<a href="mailto:Matt.Slaga@us.didata.com" target="_blank">Matt.Slaga@us.didata.com</a>><br>
>><br>>> If you are not using a cube, you might need an enhanced IOS software MTP<br>>> termination point which will allow the capturing of DTMF packets inband and<br>>> process them out of band. These can run as software MTPs on the gateway<br>
>> (cannot use the UCM built in software MTPs for this purpose).<br>>><br>>> ________________________________________<br>>> From: <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a><br>
>> [<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Nick Matthews<br>>> [<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>]<br>
>> Sent: Tuesday, July 13, 2010 11:38 AM<br>>> To: Mark Holloway<br>>> Cc: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>; Paul van den IJssel<br>>> Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue<br>
>><br>>> Also check the ccn subsystem sip on the CUE - make sure you're<br>>> configured for the DTMF method you're using everywhere else. Plus use<br>>> a CUBE as stated.<br>>><br>>> -nick<br>
>><br>>> On Tue, Jul 13, 2010 at 11:17 AM, Mark Holloway <<a href="mailto:mh@markholloway.com" target="_blank">mh@markholloway.com</a>><br>>> wrote:<br>>> > The Genband is probably doing the same thing that CUBE would be doing.<br>
>> > What<br>>> > are the DTMF requirements from your provider? Have you confirmed that<br>>> > DTMF<br>>> > is being passed to the provider? A Wireshark capture on the public side<br>
>> > of<br>>> > the S3 would verify DTMF is being passed.<br>>> > On Jul 13, 2010, at 7:58 AM, Bill wrote:<br>>> ><br>>> > You should not trunk directly to CUCM from your ITSP. You need to go<br>
>> > through<br>>> > a CUBE router. Terminate the ITSP on the CUBE and then do a SIP trunk to<br>>> > your CUBE router.<br>>> ><br>>> > ________________________________<br>>> > From: <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a><br>
>> > [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] On<br>>> > Behalf Of Paul van den IJssel<br>>> > Sent: Tuesday, July 13, 2010 9:54 AM<br>
>> > To: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>>> > Subject: [cisco-voip] SIP Trunk from ISP DTMF issue<br>>> ><br>>> > Hi all,<br>
>> ><br>>> > Currently I'm running a acceptance test on a SIP Trunk from our ISP.<br>>> > Everything is working fine except the DTMF.<br>>> ><br>>> > Setup:<br>>> > [GenBand SBC] ---- [SIP Trunk into CUCM 7.1] ---- [Cisco Unity Express<br>
>> > 7.0]<br>>> ><br>>> > The SIP Trunk as well as the CUE are in the same region supporting only<br>>> > G.711. On the SIP Trunk I've tried all different kind of DTMF<br>>> > configurations<br>
>> > (No Preference, RFC 2833 and OOB). We've tried to force the GenBand SBC<br>>> > to<br>>> > only use RFC 2833 as well as the SIP Trunk. But this didn't work, not<br>>> > even<br>
>> > after we added a MRGL with MTP's.<br>>> ><br>>> > Is there some sort of best practive to implement DTMF over a SIP Trunk?<br>>> > Is<br>>> > there a way I can do some debugging on the CUCM/CUE?<br>
>> ><br>>> > Kind regards,<br>>> ><br>>> > Paul van den IJssel<br>>> > Digacom<br>>> > _______________________________________________<br>>> > cisco-voip mailing list<br>
>> > <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>>> > <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
>> ><br>>> ><br>>> > _______________________________________________<br>>> > cisco-voip mailing list<br>>> > <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
>> > <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>>> ><br>>> ><br>>><br>>> _______________________________________________<br>
>> cisco-voip mailing list<br>>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
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