<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.25in 1.0in 1.25in;}
div.WordSection1
{page:WordSection1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=WordSection1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Dear Paul,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> Could you please clarify where exactly the “</span>"Accept
Unsolicited Notification" box of the SIP Trunk Profile” <span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>as I can’t
find this option in the sip trunk page or in the Standard Sip Profile page.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>Best
Regards<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ahmed Abd EL-Rahman</span></b><b><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#948A54'><o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Paul van den IJssel<br>
<b>Sent:</b> Friday, July 16, 2010 11:09 AM<br>
<b>To:</b> cisco-voip@puck.nether.net"<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk from ISP DTMF issue<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>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.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>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?<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Paul<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><br>
<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>2010/7/15 Nick Matthews <<a
href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>><o:p></o:p></p>
<p class=MsoNormal>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>
<span style='color:#888888'><br>
-nick</span><o:p></o:p></p>
<div>
<div>
<p class=MsoNormal><br>
On Thu, Jul 15, 2010 at 4:58 AM, Paul van den IJssel <<a
href="mailto:pijssel@gmail.com">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">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">cisco-voip-bounces@puck.nether.net</a><br>
>> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">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">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">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">cisco-voip-bounces@puck.nether.net</a><br>
>> [<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>]
On Behalf Of Nick Matthews<br>
>> [<a href="mailto:matthnick@gmail.com">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">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">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">cisco-voip-bounces@puck.nether.net</a><br>
>> > [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">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">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">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">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">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>
>> Disclaimer:<br>
>><br>
>> This e-mail communication and any attachments may contain<br>
>> confidential and privileged information and is for use by the<br>
>> designated addressee(s) named above only. If you are not the<br>
>> intended addressee, you are hereby notified that you have received<br>
>> this communication in error and that any use or reproduction of<br>
>> this email or its contents is strictly prohibited and may be<br>
>> unlawful. If you have received this communication in error,
please<br>
>> notify us immediately by replying to this message and deleting it<br>
>> from your computer. Thank you.<br>
>><br>
>><br>
><br>
> _______________________________________________<br>
> cisco-voip mailing list<br>
> <a href="mailto:cisco-voip@puck.nether.net">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>
><o:p></o:p></p>
</div>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
</body>
</html>