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<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Also check the amount of DSP’s in de router.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What happens if you enter the controller timeslot command again?<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Ryan Ratliff [mailto:rratliff@cisco.com] <br>
<b>Sent:</b> vrijdag 18 juni 2010 17:19<br>
<b>To:</b> Damian Turburville<br>
<b>Cc:</b> cips; cisco-voip mailinglist<br>
<b>Subject:</b> Re: [cisco-voip] ISDX - Call Manager transfer issue<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>With MGCP gateways you don't typically need to transcode.
Failing to allocate an MOH server due to codec mismatch should just you
tone on hold, not drop the call.<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>You'll need to look at the ccm traces to see what is going
on.<o:p></o:p></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";
color:black'>-Ryan<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>On Jun 18, 2010, at 11:12 AM, Damian Turburville wrote:<o:p></o:p></p>
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<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
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<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>Ryan,<br>
No, pressing hold cuts the call off as well.<br>
When you say a "media negotiation failure" would this be a
transcoding or MTP issue do you think?<br>
Thanks,<br>
Damian<o:p></o:p></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
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style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
<hr size=1 width="100%" align=center>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Ryan Ratliff <<a
href="mailto:rratliff@cisco.com">rratliff@cisco.com</a>><br>
<b>To:</b><span class=apple-converted-space> </span>Damian Turburville
<<a href="mailto:d_turburville@yahoo.com">d_turburville@yahoo.com</a>><br>
<b>Cc:</b><span class=apple-converted-space> </span>cips <<a
href="mailto:cisco@cips.nl">cisco@cips.nl</a>>; cisco-voip mailinglist <<a
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Sent:</b><span class=apple-converted-space> </span>Fri, June 18, 2010
2:58:44 PM<br>
<b>Subject:</b><span class=apple-converted-space> </span>Re: [cisco-voip]
ISDX - Call Manager transfer issue<br>
</span><br>
Can they put the call on hold without using the transfer feature? When
the first transfer button is pressed all that happens is the call is placed on
hold.<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>Your debug shows CCM is sending the disconnect with cause
"resources unavailable, unspecified". This is going to be a
media negotiation failure.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif"'>-Ryan<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>On Jun 18, 2010, at 8:36 AM, Damian Turburville wrote:<o:p></o:p></p>
</div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
<div>
<div>
<div>
<p class=MsoNormal><span style='font-family:"Times","serif"'>Thanks for looking
at this.<br>
<br>
The call is cut off as soon as the transfer button is pressed for the first
time so they do not get to enter any digits.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-family:"Times","serif"'><br>
Please see attached files for debugs and screenshots.<br>
<br>
Cheers,<br>
Damian<o:p></o:p></span></p>
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style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
<hr size=1 width="100%" align=center>
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<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span
class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>cips <<a
href="mailto:cisco@cips.nl" target="_blank">cisco@cips.nl</a>><br>
<b>To:</b><span class=apple-converted-space> </span>Damian Turburville <<a
href="mailto:d_turburville@yahoo.com" target="_blank">d_turburville@yahoo.com</a>><br>
<b>Cc:</b><span class=apple-converted-space> </span>cisco-voip mailinglist
<<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Sent:</b><span class=apple-converted-space> </span>Fri, June 18, 2010
1:19:45 PM<br>
<b>Subject:</b><span class=apple-converted-space> </span>RE: [cisco-voip]
ISDX - Call Manager transfer issue</span><span style='font-family:"Times","serif"'><o:p></o:p></span></p>
<div>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Also make a call and, do the transfer while running a debug isdn
q921 and debug isdn q931.</span><o:p></o:p></p>
</div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Is the call cut-off as soon as the transfer button is presses by
the user, for the first time? Or is the call cut-off when the user
presses the transfer button for the second time, to really make the transfer?</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Where does the user do a transfer to? Internal external? Or back
to the ISDX?</span><o:p></o:p></p>
</div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Please attach a screenshot of the GW configuration in the CM.</span><o:p></o:p></p>
</div>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Arial","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Arial","sans-serif"'>From:</span></b><span class=apple-converted-space><span
lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'> </span></span><span
lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'><a
href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a><span
class=apple-converted-space> </span>[mailto:cisco-voip-bounces@puck.nether.net]<span
class=apple-converted-space> </span><b>On Behalf Of<span
class=apple-converted-space> </span></b>Damian Turburville<br>
<b>Sent:</b><span class=apple-converted-space> </span>vrijdag 18 juni 2010
13:48<br>
<b>To:</b><span class=apple-converted-space> </span><a
href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b><span class=apple-converted-space> </span>[cisco-voip] ISDX
- Call Manager transfer issue</span><o:p></o:p></p>
</div>
</div>
</div>
<p class=MsoNormal> <o:p></o:p></p>
<div>
<div>
<p class=MsoNormal>Hi,<br>
Hopefully one of you clever people out there can help me out with a problem we
are having.<br>
The setup is as follows<br>
<br>
Siemens ISDX<br>
<span class=apple-converted-space> </span><i>sending
DPNSS to</i><br>
Westell protocol Convertor<br>
<span class=apple-converted-space> </span><i>converting
to qsig and sending to</i><br>
Cisco 2651XM MGCP gateway<o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal> <span class=apple-converted-space> </span><i>connected
across WAN to<span class=apple-converted-space> </span><br>
</i>Cisco Call Manager 4.1(2)<br>
<br>
The problem is that when someone on the isdx places a call to a cisco phone on
the CCM 4.1 and the cisco phone (7912 or 7960 series) attempts to press the
transfer button the call is cut off.<br>
We have run a "debug MGCP events" on the gateway but are having
trouble interpreting it, if anyone can lend a hand it would be greatly
appreciated.<br>
<br>
I have attached text files with the MGCP debug output along with the gateway
config and the output of the show version command. If there is anything else
that may help please let me know.<br>
<br>
Thanks,<br>
Damian<o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif"'><br>
<2651xm debug isdn q921.txt><2651xm debug isdn q931.txt><CCM
gateway screenshots.doc>_______________________________________________<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><o:p></o:p></span></p>
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