[asterisk-users] Page() command
Will the Page() command handle 200 SIP devices? How much time does it take for ALL 200 devices to be ready to receive audio? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page Command
At the current moment there is no way. You would need to specify all phones. If you were using real time you can write an agi that would fetch a list of all phones and then page them all. - Original Message - From: Anciso, Roy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, November 17, 2007 9:05 PM Subject: [asterisk-users] Page Command Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page Command
At 23:16 11/19/2007, Dovid B wrote: At the current moment there is no way. You would need to specify all phones. If you were using real time you can write an agi that would fetch a list of all phones and then page them all. There seems to be one here: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config - Original Message - From: mailto:[EMAIL PROTECTED]Anciso, Roy To: mailto:asterisk-users@lists.digium.comAsterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, November 17, 2007 9:05 PM Subject: [asterisk-users] Page Command Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Page Command
Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Page() command and file playback
Hey everybody,
I've been playing around with the Page() command. I'm looking for a way
to play back a pre-recorded page to the phones. I've seen, in the
archives that Doug Garstang was looking to do something similar, and the
response to his query was to use Local channels. After trying that, I
am able to play back the sound file to the 2 Cisco 7940s with
auto-answer turned on. The problem seems to be that they never leave
the conference after the sound file completes.
I'm currently running Asterisk 1.2.9.1 I've done the following:
Set up extension 18 to dial for the test, on hangup, via the 'h'
extension, run the page command.
Code below:
[from-sip]
exten = 18,1,Playback(goodbye)
esten = 18,n,NoOP(Played it back)
exten = 18,n,Hangup()
exten = h,1,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]|q)
[local-page]
exten = _X.,1,Set(TIMEOUT(absolute)=10)
exten = _X.,n,Set(CALLERID(Name)=Paging)
exten = _X.,n,Set(CALLERID(Number)=18)
exten = _X.,n,SetVar(_ALERT_INFO=Ring Answer)
exten = _X.,n,Dial(SIP/${EXTEN}||A(out))
exten = _X.,n,Hangup()
Even with the Hangup() on extension 18, the phone doesn't hangup.
Manually hanging up the phone before the page has completed, leaves the
phones in the conference along with the phone that I called 18 from.
Waiting for the page to complete and then hanging up the phone, all
participants exit the conference and the phones hangup.
It would be nice if the page() had an option to play back a file.
Any suggestions?
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [Asterisk-Users] Page() command and file playback
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Page command in 1.2
I am trying to test the Page command in Asterisk 1.2 but there seems to be a problem. I have an Aastra 480i that is set to auto answer. I have tested this in the following manner and it works: -- Executing Set(SIP/2001-1fee, ALERT_INFO=Ring Answer) in new stack -- Executing Set(SIP/2001-1fee, _ALERT_INFO=info=alert-autoanswer) in new stack -- Executing Dial(SIP/2001-1fee, SIP/1003|12|Ttr) in new stack -- Called 1003 -- SIP/1003-d8b5 is ringing -- SIP/1003-d8b5 answered SIP/2001-1fee -- Attempting native bridge of SIP/2001-1fee and SIP/1003-d8b5 == Spawn extension (casa, *551003, 3) exited non-zero on 'SIP/2001-1fee' But when I try to use the Page command with that extension it never answers: -- Executing Set(SIP/2001-8e6f, ALERT_INFO=Ring Answer) in new stack -- Executing Set(SIP/2001-8e6f, _ALERT_INFO=info=alert-autoanswer) in new stack -- Executing Page(SIP/2001-8e6f, SIP/1003|d) in new stack -- Playing 'beep' (language 'es') -- Created MeetMe conference 1023 for conference '293372787d' -- Hungup 'Zap/pseudo-1750777497' I can hear the phone ring but it never answers. Does the page command change the Alert_Info variable? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
