Yes it's normal.<br><br>Conference is like mixed "seperate calls". Your conference region is g711, and CUCM tells inside phone to start g711 stream. The other calls are like g729 streams.<br><br><a href="http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/srs/srsinter/moh.pdf">http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/srs/srsinter/moh.pdf</a><br>
<br>from <a href="http://cciev.wordpress.com">http://cciev.wordpress.com</a> :<br><p><u><b>C. G711 multicast everywhere, remote site uses MoH from local SRST flash</b></u></p>
<p>1. Create region for MoH. G711 with all other regions.</p>
<p>2. Enable multicast in audio source, server, MRG. Set hop count to 1.</p>
<p>3. Enable only G711 in service parameter (Default)<br>
4. Enable multicast in HQ and Remote LAN infrastructure (igmp snooping
in switches, pim sparse-dense in routers), Disable multicast in WAN
infrastructure.</p>
<p>5. Enable SRST MOH feature from flash.</p>
<p>6. Enable loopback interface in SRST router for PSTN users to hear MoH.</p><p><br></p><p>-<br></p>Dew Swen<br>
<br><br><div class="gmail_quote">On Wed, Dec 9, 2009 at 11:42 PM, <span dir="ltr"><<a href="mailto:steve.siltman@assurant.com">steve.siltman@assurant.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br><font face="sans-serif" size="2">Thanks for the tips!</font>
<br>
<br><font face="sans-serif" size="2">I can see that the phone is sending
the correct 239.1.1.1 address and port. The internal phones are now
hearing the music but the quality is beyond bad. I see on the same
page that the sender codec is g.722 and the recr codec is g.711u. Could
this be the issue?</font>
<br><div class="im"><font face="sans-serif" size="2"><br>
Steve Siltman<br>
Assurant Corporate Technology<br>
Senior Network Engineer - Cisco CCVP<br>
Work: 651-361-4752<br>
Cell: 651-336-5563<br>
<a href="mailto:steve.siltman@assurant.com" target="_blank">steve.siltman@assurant.com</a></font>
<br>
<br>
<br>
</div><table width="100%">
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<td width="40%"><font face="sans-serif" size="1"><b>Ryan Ratliff <<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></b>
</font>
<p><font face="sans-serif" size="1">2009-12-08 20:11</font>
</p></td><td width="59%">
<table width="100%">
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<td>
<div align="right"><font face="sans-serif" size="1">To</font></div>
</td><td><font face="sans-serif" size="1"><a href="mailto:steve.siltman@assurant.com" target="_blank">steve.siltman@assurant.com</a></font>
</td></tr><tr valign="top">
<td>
<div align="right"><font face="sans-serif" size="1">cc</font></div>
</td><td><div class="im"><font face="sans-serif" size="1"><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a></font>
</div><div class="im"></div></td></tr><tr valign="top">
<td>
<div align="right"><font face="sans-serif" size="1">Subject</font></div>
</td><td><font face="sans-serif" size="1">Re: [cisco-voip] MOH from Router Flash
between internal callers</font></td></tr></tbody></table>
<br>
<table>
<tbody><tr valign="top">
<td>
</td><td></td></tr></tbody></table>
<br></td></tr></tbody></table>
<br>
<br><div><div></div><div class="h5">
<br><font size="3">I don't recall this ever not being supported, in fact
the whole point of using the router flash for MOH was to get it working
to phones and PSTN gateways at the remote site.</font>
<br>
<br><font size="3">It is most definitely supported and there are quite a
few people that use it with no issues. The reason you were asked
for traces is because if the configuration looks good, then the traces
are the best way to confirm what MOH audio source/server CUCM is selecting
and why the multicast address the phone is told to listen to doesn't line
up with what the router is sending.</font>
<br>
<br><font size="3">You are correct and it is fairly straight forward. One
slight correction is that it is not the phone initiating the hold but rather
the CUCM server that instructs the held party to listen to the MOH stream.
The address used will be a combination of the user hold audio source
configured on the holding phone and the MRGL of the held phone.</font>
<br>
<br><font size="3">You can also find out the address the phone is being told
to listen to by putting the phone on hold, and then pointing a web browser
at the held phone's IP address. Click on the Streaming Statistics
links until you see the multicast address. If everything is configured
correctly then this address and port will match that configured on the
SRST router. If it doesn't match, then your CUCM configuration is
not correct. If it does match and the phone is still hearing silence
on hold then you need to troubleshoot the layer 2 and 3 multicast between
the SRST router and the IP phone.</font>
<br>
<br><font size="3">-Ryan</font>
<br>
<br><font size="3">On Dec 8, 2009, at 4:22 PM, </font><a href="mailto:steve.siltman@assurant.com" target="_blank"><font color="blue" size="3"><u>steve.siltman@assurant.com</u></font></a><font size="3">
wrote:</font>
<br><font size="3"><br>
</font><font face="sans-serif" size="2"><br>
Has anyone been able to get this too work? I turned on Multicast
within the internal network at this remote site and I'm unable to get the
phones to hear the MOH between two internal Cisco IP Phones. Years
ago, I read that this wasn't possible but was hoping this was resolved
in later versions. We are running v7 Call Manager with 12.4.24 router
code. MOH works fine to external customers. It seems fairly
straight forward as the one Cisco IP phone issues the Hold and sends the
multicast ip address to the other phone. I'm not sure why the other
phone doesn't pick up on the stream. Grr</font><font size="3"> <br>
</font><font face="sans-serif" size="2"><br>
If this isn't supported then I'll have to allow MOH across the WAN and
forget about using the routers flash outside of SRST.</font><font size="3">
<br>
</font><font face="sans-serif" size="2"><br>
I've got a ticket open with Cisco and they verified the configuration and
have asked for trace files. Either this TAC person doesn't know it
isn't supported or something has changed since my Cisco Voice Gateways
book was printed in 2006.</font><font size="3"> <br>
</font><font face="sans-serif" size="2"><br>
Thanks,</font><font size="3"> </font><font face="sans-serif" size="2"><br>
<br>
-Steve</font><font size="3"> <br>
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