<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>You might also check with www.star2star.com (Star2Star Communications). We did a call park, pickup, and transfer module with similar functionality. Integrates very nicely.<br><br>Justin Newman<br>nt_jnewman at yahoo.com<br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Jeff LaCoursiere <jeff@jeff.net><br><b><span style="font-weight: bold;">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Sent:</span></b> Thursday, February 19, 2009 7:02:58 AM<br><b><span style="font-weight:
bold;">Subject:</span></b> Re: [asterisk-users] call picking and transfers<br></font><br>
<br><br>On Wed, 11 Feb 2009, Philipp Kempgen wrote:<br><br>> Jeff LaCoursiere schrieb:<br>>> Working on some niche requests from one of my hotel clients. asterisk<br>>> 1.4.20-1 on CentOS, Polycom 501s.<br>>><br>>> The first request is for the Polycom's screen to show the CID of the<br>>> inbound caller when a call pick is executed, so the picker knows if the<br>>> call is internal or external. I have already "worked around" this issue<br>>> by using ALERT info to give seperate ring tones for outside and inside,<br>>> but they are used to their old Nortel switch which apparently showed the<br>>> CID immediately after the pick, and they then knew how to answer the<br>>> phone.<br>>><br>>> The second is to show CID information on the screen when a call has been<br>>> answered by the front desk, then a blind transfer sent to an internal<br>>> phone.
Today they simply see "Front Desk" and there is no indication of<br>>> who the actual caller is to distinguish internal staff, internal guest<br>>> room, or outside caller.<br>>><br>>> Has anyone attacked these things with Polycom that might share their<br>>> approach?<br>><br>> These bugs cover the functionality you need I guess:<br>><br>> <a href="http://bugs.digium.com/view.php?id=5014" target="_blank">http://bugs.digium.com/view.php?id=5014</a><br>> <a href="http://bugs.digium.com/view.php?id=13827" target="_blank">http://bugs.digium.com/view.php?id=13827</a><br>> <a href="http://bugs.digium.com/view.php?id=8824" target="_blank">http://bugs.digium.com/view.php?id=8824</a><br>><br>> However none of the patches are likely to be merged into 1.4.<br>><br><br>Hi,<br><br>I am very happy to announce that two patches:<br><br><a href="http://bugs.digium.com/view.php?id=8824"
target="_blank">http://bugs.digium.com/view.php?id=8824</a><br><a href="http://bugs.digium.com/view.php?id=14206" target="_blank">http://bugs.digium.com/view.php?id=14206</a><br><br>applied to 1.4.23.1 work perfectly on Polycom, Cisco, and Linksys phones <br>to supply:<br><br>* CALLED party information showing in display as remote phone is ringing<br>* blind and attended transfers show the original callers ID info to the <br>receiving extension's display<br>* call picks using **ext or *8 show the picked caller's ID info in the <br>display when it is completed.<br><br>These features are found in PBX equipment going back at least a decade, so <br>I am very happy to see them finally in asterisk. It does seem that these <br>patches will become part of the 1.6 release soon if not already.<br><br>Cheers,<br><br>j<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com"
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