Christopher-<br><br>Nothing like defining a complicated environment. I do have some experience in this arena- but unfortunately, not with the OH323 driver- I generally stick to the Nufone driver, as I find it more reliable overall. YMMV. One thing that might help is if you could tell us if it ever worked, or if this is a new problem that's cropped up since a particular change.
<br><br>Still- there are two areas to check- one, I'd start up some debugs on your gatekeeper, to see if the call is being signalled properly to Asterisk, and on Asterisk itself, to see what's being passed in. This is critical- I'm betting this is a simple case of dialplan mangling- but only the debug logs will tell.
<br><br>Secondly, I know that in the CCM trunk definitions, it's important to ensure the trunk definition follows the recommendations *exactly*. A big killer here is the 'Media Termination Point Required' box- generally, for transfers, you need one- and it needs to be functional. Your CCM administrator should be able to verify that it's defined and working- but CCM is not a simple setup. (I've done it, a couple of times). There are also version differences- you don't mention what release of CCM you're running, for example. CCM doesn't always comply with 'the rules' of
H.323 as we know them- they are known to do some non-standard things, and various channel drivers can take offence with that. If SIP is available (CCM 4.0 and above), you may want to consider re-architecting to it, as a good many of these problems go away under SIP.
<br><br>Let me know if I can be of further assistance.<br>-Paul Davidson<br> PlanCommunications, LLC.<br><br><div><span class="gmail_quote">On 1/12/06, <b class="gmail_sendername"><a href="mailto:asterisk-users-request@lists.digium.com">
asterisk-users-request@lists.digium.com</a></b> <<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>Message: 4<br>Date: Thu, 12 Jan 2006 11:01:02 -0500<br>From: "Peckham, Christopher" <<a href="mailto:CPeckham@mercy.edu">CPeckham@mercy.edu</a>><br>Subject: [Asterisk-Users] Transfer issue with a Cisco CCM/phone
<br>To: <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>Message-ID:<br> <<a href="mailto:52AF58BB5A49BF46AE645C9AB714F789055CDF98@exchange3.mercy.local">52AF58BB5A49BF46AE645C9AB714F789055CDF98@exchange3.mercy.local
</a>><br>Content-Type: text/plain; charset="US-ASCII"<br><br>Hello,<br><br>We have a mixed environment here consisting of a number of Avaya PBX<br>systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco
<br>router, and an Asterisk server. The PBX land is connected to the VoIP<br>land using the Cisco router/H.323 gateway.<br><br>The Asterisk system is running code from the CVS tree from around mid<br>Oct 2005. The OH323 driver on the system is from inaccessnetworks,
<br>version 0.7.3.<br><br>We are having an issue transferring calls from one system to another.<br><br>When the transfer is attempted on various sets, it does not work and the<br>call is 'lost'. When the called party is on a cisco phone, the Avaya
<br>(calling party) hears music on hold from the cisco system while the<br>transfer is being made. The Cisco user reaches the auto-attendant on<br>the Asterisk box and the music on hold is heard by the Avaya caller.<br>When the Cisco user attempts to complete the transfer by pressing the
<br>transfer button again, the music goes away on the Avaya phone but the<br>call remains on the Cisco phone and the transfer button remains active<br>on the cisco set and does not transfer the call.<br><br>Additional examples:
<br><br>THESE DO NOT WORK...<br><br>* Avaya user calls a Cisco user and talks. The called party (Cisco user)<br>then attempts to transfer to an auto-attendant on the Asterisk system.<br>The call does not transfer.<br><br>
* Cisco user calls another Cisco user and talks. The called party then<br>attempts to transfer to an auto-attendant on the Asterisk system. The<br>call does not transfer.<br><br>* Outside user calls number which passes through Avaya PBX and Cisco
<br>router to the Auto attendant. The user then dials an extension to a<br>Cisco phone. The called party on the Cisco phone can not transfer a<br>call.<br><br>THESE WORK...<br><br>* Avaya user calls another Avaya user and talks. The called party then
<br>attempts to transfers to an auto-attendant on the Asterisk system. The<br>call DOES TRANSFER.<br><br>* Cisco user calls an Avaya user and talks. The called party (Avaya)<br>then attempts to transfer to an auto-attendant on the Asterisk system.
<br>The call DOES TRANSFER.<br><br>* Outside user calls number which passes through Avaya PBX and Cisco<br>router to the Auto attendant. The user then dials an extension to an<br>Avaya phone. THE CALLED PARTY CAN TRANSFER THE CALL.
<br><br>Any assistance in solving this issue would be appreciated.<br><a href="mailto:cpeckham@mercy.edu">cpeckham@mercy.edu</a><br><br><br>-</blockquote></div><br>