My SIP registration are name sort of like this : phonea-exten1, phone1-exten2, etc. Makes it easy to loop, I can send you a snippet tomorrow. But you have to know in advance all the SIP peer names.
Mike From: [email protected] [mailto:[email protected]] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] advice re: Page() application Thanks Mike - this does help. The setup will be a local server on the LAN, and hopefully have plenty of snort to handle the load (20-30 phones). I also am not quite ready to put out 1.8 for my users yet. Do you have a snippet of dialplan code you'd be willing to share to loop through a group and grab/build up a list of channels as you describe? That would be enlightening (and probably save me some time)! What I am hearing is - using a second line presence for the Page() function will work; auto-answer should work and I should only page the phones that are not in use. Cassius -------- Original Message -------- Subject: Re: [asterisk-users] advice re: Page() application From: "Mike" < <mailto:[email protected]> [email protected]> Date: Thu, October 14, 2010 10:12 am To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" < <mailto:[email protected]> [email protected]> Hi Cassius, Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the Wiki and you`ll do good. Two warnings: It seems to me that the adhoc MeetMe room used by the page application slows things down quite a lot. If you page and have a phone nearby, you`ll hear yourself with quite a bit of delay. It`s very annoying if you`re paging and hearing the page at the same time. Apparently 1.8 supports multicast and will do this differently, but it’ll be a long while before I trust 1.8 to be stable enough for my needs. If you`re doing this over an Internet link (i.e. hosted PBX), keep in mind that because of the MeetMe (I imagine), even if the receiving phones aren’t creating audio, the bandwidth is still is used as if everyone was talking at the same time in a MeetMe room. No biggie if everything is on the LAN, but a bit of a problem if not and you have many phones. And here is a tip: auto-answer is good, but you`ll have to loop through every SIP registration on the phone before using Page() to see if they are being used before adding them to the Page. If not, the phone will not auto-answer (since you`re on a call already) but you`ll have a missed call everytime somebody pages you while you`re on the phone. Users hate that (with reason). You check if each and every phone is being used BEFORE adding them to your page. In other words, if 10 out of 15 phones are idle, Page() only those 10. Besides that, things work as advertised. Mike From: <mailto:[email protected]> [email protected] [ <mailto:[email protected]> mailto:[email protected]] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] advice re: Page() application Hi all, I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down. To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up. I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated! Regards, Cassius Smith _____ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by <http://www.api-digital.com> http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: <http://www.asterisk.org/hello> 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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