[asterisk-users] Cisco 7960 Multiline phone

[Ishfaq Malik]

Hi

You could also do it with one extension but set the call limit for the 
extension in the sip.conf to something like
call-limit=3
Which would allow 3 concurrent calls to the one extension

Ish

Jimmy Ezell wrote:
>
> 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:* asterisk-users-bounces at lists.digium.com
>     [mailto:asterisk-users-bounces at lists.digium.com] *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:* asterisk-users-bounces at lists.digium.com
>     [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
>     *Jimmy Ezell
>     *Sent:* Tuesday, August 11, 2009 12:18 PM
>     *To:* asterisk-users at lists.digium.com
>     *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:* 'asterisk-users at lists.digium.com'
>         *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

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office: 0161 660 3062