<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)">
<!--[if !mso]>
<style>
v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
</style>
<![endif]-->
<style>
<!--
/* Font Definitions */
@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;}
@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0cm;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
color:black;}
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;}
pre
{mso-style-priority:99;
mso-style-link:"HTML Preformatted Char";
margin:0cm;
margin-bottom:.0001pt;
font-size:10.0pt;
font-family:"Courier New";
color:black;}
span.EmailStyle17
{mso-style-type:personal;
font-family:"Calibri","sans-serif";
color:windowtext;}
span.HTMLPreformattedChar
{mso-style-name:"HTML Preformatted Char";
mso-style-priority:99;
mso-style-link:"HTML Preformatted";
font-family:Consolas;
color:black;}
span.EmailStyle20
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page Section1
{size:612.0pt 792.0pt;
margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.Section1
{page:Section1;}
-->
</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 bgcolor=white lang=EN-AU link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><span style='color:#1F497D'>Hi<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>That command didn't seem to make
a difference, however I created a SIP trunk from CUCM to the 2821 router and
then from the 2821 to the GSM gateway. Now it's SIP from CUCM through to the
GSM. All the ringback works fine, consultative transfer and Adhoc conferences
are also good (no disconnects). Also no need to enable MTP on the H.323 gateway
so there are no annoying messages about video bandwidth on PSTN calls. There is
however MTP on the SIP trunk in CUCM.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Although the original problem
isn't fixed, I'm pretty sure you are on the right track with it being a H.323 compatibility
issue but taking out H.323 seems to have solved it.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>The reason for wanting to send the
calls to 2821 first was that dial peer hunting on the 2821 will allow calls to
out via my ISDN if the SIP trunk to the GSM gateway is full or not answering. I
could send the calls direct to GSM gateway from CUCM but I don't know to make CUCM
hunt to a h.323 gateway if a SIP trunk is not available (you can't add SIP trunks
to route groups can you?)<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Thanks,<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Nick.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'>From:</span></b><span lang=EN-US
style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>
Adam Frankel [mailto:afrankel@cisco.com] <br>
<b>Sent:</b> Friday, 7 August 2009 10:19 AM<br>
<b>To:</b> Mooney, Nicholas<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Adhoc conference calls drop without MTP<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>More than likely the H323 gateway is not supporting
emptycapabilities. Make sure you are on a recent IOS version and put the
'emptycapabilities' command under voice service voip->h323.
Another option would be to use an MTP which supports codec passthrough.
Keep in mind with the MTP in use, you will need enough video locations between
the Video enabled IP Phone and the MTP as well as the MTP and the H323
gateway. <br>
<br>
Adam<br>
<br>
-------- Original Message --------<br>
Subject: [cisco-voip] Adhoc conference calls drop without MTP<br>
From: Mooney, Nicholas <a href="mailto:Nicholas.Mooney@astrazeneca.com"><Nicholas.Mooney@astrazeneca.com></a><br>
To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Date: 8/6/09 6:26 PM<br>
<br>
<o:p></o:p></p>
<p class=MsoNormal>Hi<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>I have an AudioCodes GSM (SIP) gateway connected to a 2821
H.323 gateway via SIP which is then connected to CUCUM via h.323<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>Regular outbound calls work fine. CUCM sends all 0.@ calls
to the H.323 gateway and then the gateway matches mobile phones call to a SIP
dial-peer and sends the call to the AudioCodes GSM gateway.<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>The only problem is when a call via the GSM gateway gets
added to an adhoc conference, it gets dropped/disconnected all together after
about 10 seconds. If I configure the H.323 gateway to require an MTP then the
calls don’t get dropped, however any CUVA (Video Advantage) user who
makes a PSTN call gets a message on their phone saying “Video Bandwidth
Unavailable”. The CUVA doco says this is because an MTP is being used and
MTP’s don’t support video.<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>The only MTP’s in my network are the default ones
configured in CUCUM.<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>Any ideas on how I can get the calls via the GSM gateway not
to drop when added to an adhoc conference without enabling an MTP on the
gateway in CUCM? <o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>Or is there a way to have PSTN calls for CUVA users not show
the “Video Bandwidth Unavailable” message when it’s clearly
an audio-only call when an MTP is enabled for the gateway? <o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>Thanks,<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal>Nick.<o:p></o:p></p>
<pre><o:p> </o:p></pre><pre style='text-align:center'>
<hr size=4 width="90%" align=center>
</pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>cisco-voip mailing list<o:p></o:p></pre><pre><a
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><o:p></o:p></pre><pre><a
href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></pre><pre> <o:p></o:p></pre>
<p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p> </o:p></span></p>
</div>
</div>
</body>
</html>