I have already upgraded the phone load on this one phone but it appears to be happening on more than one phone now, I will have to get the users to monitor it and see which phone it sstill happens to.<br><br>
<div><span class="gmail_quote">On 07/12/2007, <b class="gmail_sendername">Jonathan Charles</b> <<a href="mailto:jonvoip@gmail.com">jonvoip@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Well, then load the new firmware files onto the TFTP server and change<br>the load for this phone, see if it helps... that would cause no
<br>downtime (other than resetting this phone).<br><br><br>Jonathan<br><br>On Dec 7, 2007 7:57 AM, Nick <<a href="mailto:csvoip@googlemail.com">csvoip@googlemail.com</a>> wrote:<br>> Would love to but we have over 8000 phones and they don't like down time.
<br>><br>><br>><br>><br>> On 07/12/2007, Jonathan Charles <<a href="mailto:jonvoip@gmail.com">jonvoip@gmail.com</a>> wrote:<br>> > Well, you may want to upgrade to 4.1(3)sr5d and get your phone loads
<br>> > updated... (the sr5d already contains the latest dev pack)...<br>> ><br>> ><br>> > Jonathan<br>> ><br>> > On Dec 7, 2007 6:06 AM, Nick <<a href="mailto:csvoip@googlemail.com">
csvoip@googlemail.com</a>> wrote:<br>> > > Wes<br>> > ><br>> > > Ok, I don't believe we have a codec mismatch we are using G711 and we do<br>> not<br>> > > use MOH, the MRGL on the device pool does not contain any MOH resources.
<br>> > ><br>> > > Is there anyway I could check to see if there anything is interupting<br>> the<br>> > > control channel during the media redirection, if the call has dropped.<br>> > >
<br>> > ><br>> > ><br>> > ><br>> > ><br>> > > On 06/12/2007, Wes Sisk <<a href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>> wrote:<br>> > > ><br>> > > > Not necessarily.
<br>> > > ><br>> > > > Hold/transfer does place the call on hold. That works fine unless one<br>> of<br>> > > the call legs meets the following criteria in 4.1:<br>> > > ><br>
> > > > 1) is h323 to a device that is not h323v2 compliant with empty<br>> > > capabilities set (ECS) support<br>> > > > 2) use the CM SIP trunk<br>> > > ><br>> > > > If any call leg uses those then MTP will be required. Otherwise,
<br>> > > generally, MTP should not be required.<br>> > > ><br>> > > > Transfers typically drop calls when:<br>> > > > 1) there is a codec mismatch between the ingress gateway and the MoH
<br>> > > server. The codec negotation failure usually puts the call in a state<br>> where<br>> > > it cannot be resumed or it just immediately drops<br>> > > > 2) there is an error closing/opening media channels. This happens
<br>> with<br>> > > H.323 endpoints that do not support ECS. It happens with SIP endpoints<br>> that<br>> > > do not handle the re-invite. It happens when there is a network<br>> condition<br>> > > that delays or interrupts the control channel (h323, sip, mgcp
<br>> signaling)<br>> > > during the media redirection.<br>> > > ><br>> > > > /Wes<br>> > > ><br>> > > ><br>> > > > Jonathan Charles wrote:<br>> > > > Because when you hit the transfer key, the call is placed on hold, to
<br>> > > > do that, you need an MTP.<br>> > > ><br>> > > ><br>> > > > Jonathan<br>> > > ><br>> > > > On Dec 6, 2007 12:53 PM, Voice Noob <<a href="mailto:voicenoob@gmail.com">
voicenoob@gmail.com</a>> wrote:<br>> > > ><br>> > > ><br>> > > > Why would he need an MTP to transfer a call?<br>> > > ><br>> > > ><br>> > > ><br>
> > > ><br>> > > > On Dec 6, 2007 12:47 PM, Jonathan Charles <<a href="mailto:jonvoip@gmail.com">jonvoip@gmail.com</a>> wrote:<br>> > > ><br>> > > ><br>> > > >
<br>> > > > Is there an MRGL assigned to the device pool? Cuz you need an MTP to<br>> > > ><br>> > > > transfer...<br>> > > ><br>> > > ><br>> > > > Jonathan
<br>> > > ><br>> > > ><br>> > > ><br>> > > ><br>> > > > On Dec 6, 2007 10:59 AM, Nick < <a href="mailto:csvoip@googlemail.com">csvoip@googlemail.com</a>> wrote:
<br>> > > ><br>> > > ><br>> > > > Has anyone come across an issue before where when you press the<br>> transfer<br>> > > > button the call gets dropped, I have one user who claims that when one
<br>> > > > particular cell phone number calls in and she tried to transfer she<br>> > > ><br>> > > > loses<br>> > > ><br>> > > ><br>> > > ><br>> > > > the call.
<br>> > > ><br>> > > > This is on a 7960 Version: 7.2 (4.0)<br>> > > ><br>> > > > CCM 4.1(3) sr3c<br>> > > ><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">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">
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">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">
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<br>> ><br>><br>><br></blockquote></div><br>