Patrick M. Gray, Jr.
2005-May-02 10:52 UTC
[Asterisk-Users] 7960 "multi-line" configuration
I'm trying to configure a 7960 in a small-office setting to function like the large avaya-type PBX systems I have used. Basically each station has a DID number, and 4 or more "lines" on the phone. If you're on "line 1" and someone calls, "line 2" rings rather than a call waiting beep. Similarly you can conference two lines, etc. In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should automatically roll calls to a vacant line. When I configure the 7960 in this manner, any second call to a given DID just goes to voicemail. Is this the "best practice" way to configure the 7960, or should I assign 6 different lines to the 7960 and create a call group? Thanks! Pat
Patrick M. Gray, Jr. wrote:> In google'ing around a bit, it seems I should be able to assign the same > extension to several of the SIP lines on the 7960, and asterisk shouldI don't think that is possible, at least not the way one thinks it would work. I have also done some reading on this, maybe this thread gives a solution: http://lists.digium.com/pipermail/asterisk-users/2004-March/039271.html But, I am also curious on how other people have solved this, especially with using AMP for example. Cheers..
Gregory Wiktor - ADCom Corp.
2005-May-03 01:12 UTC
[Asterisk-Users] 7960 "multi-line" configuration
I setup my 7960 with line 1 as main, and 2 as a queue line. So if the line is busy, asterisk queues the call and it will continue to ring on line 2. Call waiting works too, but not as well as queueing... Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joris Vandalon Sent: Tuesday, May 03, 2005 2:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration On Mon, 2005-05-02 at 16:54 -0400, Patrick M. Gray, Jr. wrote:> No errors, asterisk just immediately sends the other call to voicemail> if there is already a call in progress.Try turning on "Call waiting" on your cisco phone. Cheers, Joris _______________________________________________ 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
Gregory Wiktor - ADCom Corp.
2005-May-03 01:16 UTC
[Asterisk-Users] 7960 "multi-line" configuration
I actually setup 6 registrations as separate lines. This allows me dialout selection, like line 5 for teliax, line 6 for voipjet, etc. I suspect you need different logons or all of your lines would ring at once. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris A. Icide Sent: Tuesday, May 03, 2005 1:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration -----BEGIN PGP SIGNED MESSAGE----- On 09:50 PM 5/2/2005, Matthew Boehm wrote: > >Hold up. So you have Phone #1. And all 6 lines register with the username of >"phone1" ? > >And you have phone #2; and all 6 lines register with username of "phone2"? > >And the phone only registers once? Interesting..I'm gonna test this. Sounds >like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris -----BEGIN PGP SIGNATURE----- Version: PGP 8.1 iQCVAwUBQncHJ+0LTNca2q41AQHf/QP7BB5ni6GOzc7JxvavF+ryg172gBtlIWku hmw5JkcinUBGKcRQ9paMXcZ+NRMokUFyljF+Yc1xLWPp4Gt1u/PCYmnU2tO/RIYg JAffPN5fVcA5zq5+uw/n0utwpUpo0VTzKPErcHonLJrr+ZF7MYxIiQ3NxHpQjAeR zOZw81xfyvU=vHoj -----END PGP SIGNATURE----- _______________________________________________ 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
Eric Wieling aka ManxPower wrote:> Correct. Asterisk does not support multiple logins from the same SIP > username. HOWEVER, if you configure the same SIP username/secret on > all 6 lines of a Cisco (or Polycom) the PHONE will NOT register using > the same username/secret more than once. It will see that more than > one line is using the same username/secret and only register once. > This is specific to the phone in question.Polycoms don't seem to handle this very well. (IP 500, IP 600). First, you can't disable call waiting, so each line has two call appearances. If I configure line 1 and line 2 with the same SIP username: I can dial out from call appearance 1 and call appearance 2 on line 1, And at the same time: I can dial out from call appearance 1 and call appearance 2 on line 2. That's all well and good. But when both call appearances on line 1 are busy the phone rejects new incoming calls - it doesn't roll down to line 2. Which means this isn't a useful way to use a Polycom as a receptionist phone. -- No virus found in this outgoing message. http://www.avg-antivirus.net/ Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.2 - Release Date: 5/2/2005