I'm with Cristobal and am wondering if there's not a routing issue between the remote IPC devices and the local gateway LAN address (or mgcp bound interface). <br><br>With transfer, the audio stream is between the local gateway and remote IPC device. With conference, the audio stream is between the UCM server and remote IPC device. Is it possible that the conferences are going through the software bridge on the UCM server as g711 conferences?<br>
<br><div class="gmail_quote">On Wed, Nov 3, 2010 at 3:47 PM, Cristobal Priego <span dir="ltr"><<a href="mailto:cristobalpriego@gmail.com">cristobalpriego@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
These are my 2 cents on your issue<br><br>when the transfer softkey is pressed or the conferene softkey is pressed, ccm intructs both phones to "close receive channel & stop media transmission. It Stops the RTP 2-way audio Stream<br>
when transfer/conference is pressed the second time, CCM sends a series of Open Receive Channel message to the phones and instructs the phones to Start Media Transmission.<br><br>So my thoughts are that when the transfer is pressed the 2nd time the RTPs aren't being routed to the ipc properly and they get 1 way audio.<br>
<br><div class="gmail_quote">2010/11/3 Scott Voll <span dir="ltr"><<a href="mailto:svoll.voip@gmail.com" target="_blank">svoll.voip@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div><div></div><div class="h5">
No. the calls go out our ASA via the anyconnect client (2.5.1025) out to the IPC user.<div><br></div><div>Transcoders are not being used. (checked with one of our tickets)</div><div><br></div><div>simply press the transfer say I have X on the line, transfer.</div>
<div><br></div><div>work around ends up being conference, I have x on the line, conferenec, then hang up.</div><div><br></div><div>Technically on a transfer the call coming in is between the VGW and the (g711)operator, then a second call is placed from the operator to the(g729) IPC user. then a join is used to connect the two if I understand correctly, thus the VGW and the IPC user should negotiate the Call as G729.</div>
<div><br></div><div>maybe I'm not understanding correctly?</div><div><br></div><div><font color="#888888">Scott</font><div><div></div><div><br><br><div class="gmail_quote">On Wed, Nov 3, 2010 at 1:08 PM, <span dir="ltr"><<a href="mailto:george.hendrix@l-3com.com" target="_blank">george.hendrix@l-3com.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Sounds like a codec issue to me as well. Are the calls
transferred back out the same gateway? Are there DSP transcoders
available for the gateways and phones and UCCX servers? When you say transfer,
is that simply using their phone or the CCX application? </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="color: black;">Bill
</span><span style="color: gray;"></span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div style="border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On
Behalf Of </b>Scott Voll<br>
<b>Sent:</b> Wednesday, November 03, 2010 3:14 PM<br>
<b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> [cisco-voip] Difference between transfer and conference</span></p>
</div><div><div></div><div>
<p class="MsoNormal"> </p>
<p class="MsoNormal">as far as the call setup / call flow, what is the difference
between a transfer and conference. Forgive me if someone answered before.</p>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">we have been having on going issues with telecommuters (IP
communicator) getting calls transfered to them from our operators. </p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">Call can be transfered and we get the call connected but no
way audio (sounds like a codec issue to me But it doesn't fail.... we have to
hang up) but doing the same thing with the conference button then dropping off
the call works fine.</p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">had about 3-8 TAC cases open on this and still have the same
issue. much less, but still have the issue.</p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal">Call flow is call comes in PSTN via PRI / or CAS T1 on a
3845 running 15.0.1M3 (one TAC case had us add <span><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">mgcp
behavior g729-variants static-pt</span></span></p>
</div>
<p class="MsoNormal" style="margin-left: 0.25in;"><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">mgcp
behavior dynamically-change-codec-pt disable which helped a lot but didn't fix
the whole problem)</span></p>
<p class="MsoNormal" style="margin-left: 0.25in;"> </p>
<p class="MsoNormal" style="margin-left: 0.25in;"><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">goes
to UCCx agent (otherwise known as operator) who transfers to IP Communicator
(telecommuters). The problem we still have is that sometimes (2-10 times
a day) we get the call transfered but there is no audio. But if they
instead conference them in then exit the conference it works.</span></p>
<p class="MsoNormal" style="margin-left: 0.25in;"> </p>
<p class="MsoNormal" style="margin-left: 0.25in;"><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">other
TAC cases we have open are for IP Communicator and are working on upgrading all
IPC users to 7.0.5.1.</span></p>
<p class="MsoNormal" style="margin-left: 0.25in;"> </p>
<p class="MsoNormal" style="margin-left: 0.25in;"><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">What
would a transfer do(or not do), that a conference call doesn't?</span></p>
<p class="MsoNormal" style="margin-left: 0.25in;"> </p>
<p class="MsoNormal" style="margin-left: 0.25in;"><span>any
comments / thoughts / answers appreciated.</span></p>
<p class="MsoNormal" style="margin-left: 0.25in;"> </p>
<p class="MsoNormal" style="margin-left: 0.25in;"><span style="font-family: "Courier New"; color: rgb(31, 73, 125);">Scott</span></p>
</div></div></div>
</div>
</blockquote></div><br></div></div></div>
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