>From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup. Directed pickup is enabled by adding the following lines to extensios.conf exten => _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2}) exten => _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK) On the phone side for each line that is going to be monitored add lines like the following to the phone's cfg file. attendant.reg="1" attendant.resourceList.1.address="sip:205@192.168.1.102" attendant.resourceList.1.label="205" attendant.resourceList.2.address="sip:217@192.168.1.102" attendant.resourceList.2.label="217" call.directedCallPickupMethod="legacy" call.directedCallPickupString="*8" feature.12.name="directed-call-pickup" feature.12.enabled="1" Assuming my server is at 192.168.1.102, this will add two BLF lines to the phone for extensions 205 and 217. Calls incoming to those extensions will show a blinking green led on the monitoring phone, pressing the hard key will pick the call up, if it is answered elsewhere the led will change to solid red. AFAIK this cannot be configured via the phones web gui, you must use the cfg files. You can also use versions of Asterisk older than 1.6.1 if you remove the restriction on what asterisk thinks Polycom phones can handle. Look in chan_sip.c for if (strstr(p->useragent, "Polycom")) { p->subscribed = XPIDF_XML; and change that line to p->subscribed = DIALOG_INFO_XML; On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere <j...@sunfone.com> wrote: > > Struggling with an IP650 and 7 IP335s this morning. I have the following > hints defined (courtesy of FreePBX 2.9): > > extensions_additional.conf:**exten => 300,hint,SIP/300 > extensions_additional.conf:**exten => 301,hint,SIP/301 > extensions_additional.conf:**exten => 302,hint,SIP/302 > extensions_additional.conf:**exten => 303,hint,SIP/303 > extensions_additional.conf:**exten => 304,hint,SIP/304 > extensions_additional.conf:**exten => 305,hint,SIP/305 > extensions_additional.conf:**exten => 307,hint,SIP/307 > extensions_additional.conf:**exten => 308,hint,SIP/308 > extensions_additional.conf:**exten => 322,hint,SIP/322 > extensions_additional.conf:**exten => 350,hint,SIP/350 > extensions_additional.conf:**exten => 400,hint,SIP/400 > > The Polycoms are all pulling an XML directory via FTP where each extension > has "<BW>" (Buddy Watch) set to 1: > > <item> > <ln>Mehra</ln> > <fn>Ray</fn> > <ct>301</ct> > <sd>101</sd> > <bw>1</bw> > </item> > > This all actually works fine, and from the reception phone (the 650) I can > see the status of all the extensions, and if I dig into some menus on the > 335 I can see status as well. So I would expect that "core show hints" > would show '8' for all extensions, but it doesn't: > > artha*CLI> core show hints > > -= Registered Asterisk Dial Plan Hints =- > 300@ext-local : SIP/300 State:Idle > Watchers 7 > 301@ext-local : SIP/301 State:Idle > Watchers 8 > 302@ext-local : SIP/302 State:Idle > Watchers 8 > 303@ext-local : SIP/303 State:Idle > Watchers 8 > 304@ext-local : SIP/304 State:InUse > Watchers 8 > 305@ext-local : SIP/305 State:Idle > Watchers 7 > 307@ext-local : SIP/307 State:Idle > Watchers 1 > 308@ext-local : SIP/308 State:Idle > Watchers 7 > 350@ext-local : SIP/350 State:Idle > Watchers 1 > 400@ext-local : SIP/400 State:InUse > Watchers 7 > ---------------- > - 11 hints registered > > > Something seems broken here. And the 650 seems to "lose" its hint for a > phone once in a while, and report it as unreachable, even though it can > easily make and receive calls from it. > > Am I tilting at windmills? Is this really unstable or has someone made it > work solidly? > > Thanks! > > -- > > Jeff LaCoursiere > SunFone > 340-715-7600 x222 > j...@sunfone.com > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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