With that phone what you really probably want to do is just configure them all with the same details...
i.e. # Line 1 appearance line1_name: "incoming" line1_shortname: "Incoming (Line1)" line1_authname: "incoming" line1_password: "password" # Line 2 appearance line2_name: "incoming" line2_shortname: "Incoming (Line2)" line2_authname: "incoming" line2_password: "password" # Line 3 appearance line3_name: "incoming" line3_shortname: "incoming (Line3)" line3_authname: "incoming" line3_password: "password" # Line 4 appearance line4_name: "incoming" line4_shortname: "Incoming (Line4)" line4_authname: "incoming" line4_password: "password" # Line 5 appearance line5_name: "incoming" line5_shortname: "incoming (Line5)" line5_authname: "incoming" line5_password: "password" # Line 6 appearance line5_name: "102" line5_shortname: "Ext. 102 (Line1)" line5_authname: "102" line5_password: "password" in the phone config file... Then, in extensions.conf exten => workhours,1,Dial(SIP/incoming) exten => workhours,n,Voicemail(100,u) ... The phone will only actually register multiple times for 'incoming' though asterisk just handles that and calls to 'incoming' will come through on the lowest available line and show as call waiting with an 'Answer' soft key allowing the next call to be answered placing the current call on hold... Seems to be exactly what you want... d 2009/8/12 Jimmy Ezell <[email protected]> > Sorry for not being real clear. > > What I have is 1 front desk phone only with 6 lines > Front Desk Phone line 1 - incoming extension 1 > Front Desk Phone line 2 - incoming extension 2 > Front Desk Phone line 3 - incoming extension 3 > Front Desk Phone line 4 - incoming extension 4 > Front Desk Phone line 5 - incoming extension 5 > Front Desk Phone line 6 - inside office extension > > If incoming line 1 is busy I want the next incoming call to come in on line > 2. > If incoming line 2 and 3 are busy but 1 is free the next call should got to > line 1. > > So lines 1 and 2 might get a lot of calls but only on really busy days will > calls make it up to lines 4 and 5. > > Does that make sense? Anyone have the solution? > > > *Jimmy Ezell > * > > > ------------------------------ > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *David Gibbons > *Sent:* Tuesday, August 11, 2009 12:39 PM > > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone > > Jimmy, > > > > To clarify, you want to configure the phones like this where p means phone > and l means logical line: > > > > Phone 1: > > P1l1 > > P1l2 > > P1l3 > > > > Phone 2: > > P2l1 > > P2l2 > > P2l3 > > > > Phone 3: > > P3l1 > > P3l2 > > P3l3 > > > > It sounds like (and looks like) you’re dialing all of the extensions on one > phone at the same time, which is why they’re ringing and ringing. What you > want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1) > in the dial command to ring them simultaneously. asterisk will then fail > through if none of the phones answer in time. > > > > -Dave > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Jimmy Ezell > *Sent:* Tuesday, August 11, 2009 3:05 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone > > > > Thanks for the help, I really appreciate the feedback. > > > > I tried ringing them all at the same time as you suggested: > > exten => > workhours,1,Dial(SIP/incomming1&SIP/incomming2&SIP/incomming3&SIP/incomming4&SIP/incomming5) > > but it does very strange stuff: > > - I have to push the extension button twice to answer. > > - More then one extension shows off hook at the same time (Maybe 2 or 3 of > the 5 will show off hook on the phone) > > - When I hang up the phone starts to ring again even though there is no > caller > > > > I tried ringing them in order: > exten => workhours,1,Dial(SIP/incomming1,5,r) > exten => workhours,n,Dial(SIP/incomming2,5,r) > exten => workhours,n,Dial(SIP/incomming3,5,r) > exten => workhours,n,Dial(SIP/incomming4,5,r) > exten => workhours,n,Dial(SIP/incomming5,5,r) > > exten => workhours,n,Macro(voicemail,100) > > > > Now I see the call march along each of the extensions until it gets to the > end goes to voice mail. > > > > What I really want is for the call to go to only one of the unused lines > and then fall straight through to voicemail after the timeout. > > Anyone have some thoughts on getting it to work that way? > > > > ------------------------------ > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *David Gibbons > *Sent:* Tuesday, August 11, 2009 10:05 AM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone > > Yes each extension needs to be configured separately in the cisco CNF file. > > > > I use a distinct extension on each phone (2 phones can’t register to one > ‘extension’ afaik) and ring them in order: > > > > 1,1,Dial(SIP/xx) > > 1,n,Dial(SIP/xx1) > > 1,n,Dial(SIP/xx2) > > > > Or ring them at the same time: > > 1,1,Dial(SIP/xx&SIP/xx1&SIP/xx2) > > > > Someone else may have better solution though. > > > > -Dave > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Jimmy Ezell > *Sent:* Tuesday, August 11, 2009 12:18 PM > *To:* [email protected] > *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone > > > > Sorry I mean to say cisco 7960 phone. > > > > > ------------------------------ > > *From:* Jimmy Ezell > *Sent:* Tuesday, August 11, 2009 9:15 AM > *To:* '[email protected]' > *Subject:* Cisco 1760 Multiline phone > > I have a cisco 1760 phone running sip and I need to configure for our > receptionist so that she can answer calls on more then one extension. > > What is the easiest way to configure this so that incomming calls go to the > next availble extension? > > Does each extension on the phone need to be set seperately in the sip.conf > file (see below for my example)? > > > > sip.conf file > ================= > > [incomming1] > > type=friend > context=internal > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > mailbox=100 > > > > [incomming2] > type=friend > context=internal > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > mailbox=100 > > > > [incomming3] > type=friend > context=internal > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > mailbox=100 > > =================== > > *Jimmy Ezell** > *Assistant IT Manager > *(408) 487-2200** > * <http://www.hmhca.com/> > > > > > > * * > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
