can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050412/7b708d9c/attachment.htm
On Tuesday 12 April 2005 10:18 am, MobilPete wrote:> can anyone help ?? > trying to get Polycom IP300 to utilize both lines, would like calls to roll > to open line when incoming call arrives while user is on line 1. Looked > everywhere and tried many things with no luck.Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- Josiah Bryan IT Coordinator Productive Concepts, Inc. jbryan@productiveconcepts.com (765) 964-6009, ext. 224
we tried both, setting it as same and also seperate. but niether worked. ----- Original Message ----- From: "Josiah Bryan" <jbryan@productiveconcepts.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, April 12, 2005 9:41 AM Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300> On Tuesday 12 April 2005 10:18 am, MobilPete wrote: >> can anyone help ?? >> trying to get Polycom IP300 to utilize both lines, would like calls to >> roll >> to open line when incoming call arrives while user is on line 1. Looked >> everywhere and tried many things with no luck. > > Do you have your lines register sepratly? E.g. is there a seperate entry > in > sip.conf for each line or do they both register as the same sip device? > > > > -- > Josiah Bryan > IT Coordinator > Productive Concepts, Inc. > jbryan@productiveconcepts.com > (765) 964-6009, ext. 224 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
If you have two lines registered to one phone then you need to do the following... This assumes extensions 1001 and 1002 are your line appearances... exten => 1001,1,Dial(1001,20,trf) ;we are dialing line 1 -- After 20 seconds it will timeout and go to the next line exten => 1001,2,Dial(1002,20,trf) ;just told it to dial line 1002 exten => 1001,3,Do your voice Mail Here exten => 1001,4,Hangup You could alternately just use a GoTo after the 1st dial attempt times out and send the call to 1002 If you are talking about getting a second call while on line 1, then you just need to enable call waiting on the Asterisk box. The phone should automatically show a second incoming call and allow you to place call 1 on hold. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Josiah Bryan Sent: Tuesday, April 12, 2005 7:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 On Tuesday 12 April 2005 10:18 am, MobilPete wrote:> can anyone help ?? > trying to get Polycom IP300 to utilize both lines, would like calls to> roll to open line when incoming call arrives while user is on line 1. > Looked everywhere and tried many things with no luck.Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- Josiah Bryan IT Coordinator Productive Concepts, Inc. jbryan@productiveconcepts.com (765) 964-6009, ext. 224 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Polycom enables call waiting on each line button. If you wish the second call to go directly to the second button you need o keep track of this with group in * and control with your dial plan. On Apr 12, 2005, at 9:41 AM, Josiah Bryan wrote:> On Tuesday 12 April 2005 10:18 am, MobilPete wrote: >> can anyone help ?? >> trying to get Polycom IP300 to utilize both lines, would like calls >> to roll >> to open line when incoming call arrives while user is on line 1. >> Looked >> everywhere and tried many things with no luck. > > Do you have your lines register sepratly? E.g. is there a seperate > entry in > sip.conf for each line or do they both register as the same sip device? > > > > -- > Josiah Bryan > IT Coordinator > Productive Concepts, Inc. > jbryan@productiveconcepts.com > (765) 964-6009, ext. 224 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Noah Miller
2005-Apr-12 08:19 UTC
[Asterisk-Users] Re: multiple line usage on Polycom IP300
> On Tuesday 12 April 2005 10:18 am, MobilPete wrote: >> can anyone help ?? >> trying to get Polycom IP300 to utilize both lines, would like calls >> to roll >> to open line when incoming call arrives while user is on line 1. >> Looked >> everywhere and tried many things with no luck. > > Do you have your lines register sepratly? E.g. is there a seperate > entry in > sip.conf for each line or do they both register as the same sip device?Yes, a good way to do it is to register each line separately, like this: sip.conf [100] type=friend username=100 secret=100 callerid="100" host=dynamic dtmfmode=rfc2833 context=extensions_context mailbox=100@vm_context disallow=all allow=ulaw [100b] type=friend username=100b secret=100b callerid="100" host=dynamic dtmfmode=rfc2833 context=extensions_context disallow=all allow=ulaw Then, you can use SetGroup and CheckGroup like this in your dialplan to bypass the annoying call waiting "feature": extensions.conf exten => 100,1,SetGroup(100) exten => 100,2,CheckGroup(1) exten => 100,103,Goto(100b,1) exten => 100,3,Dial(SIP/100,20) exten => 100,4,Voicemail(su100) exten => 100,5,Hangup exten => 100b,1,Dial(SIP/100b,20) exten => 100b,2,Voicemail(sb100) exten => 100b,3,Hangup exten => 100b,102,Voicemail(sb100) exten => 100b,103,Hangup - Noah