Re: [asterisk-users] First-time queue app: verifying human member?
Matt Riddell wrote:
> Erik Anderson wrote:
>> Good evening all - for the first time, I'm implementing my first-ever
>> queue in asterisk. Overall, it's a pretty simple setup, 4 static
>> members, very low call volume, etc. The one thing that has stumped me
>> so far, though, is the following...
>
>> This is a queue I'm setting up for contacting our IT support staff
>> off-hours. As such, I've just added the cell phone numbers of our
>> staff as members. I'd like to somehow verify that it's an actual human
>> answering the phone when a member is dialed and not their mobile
>> phone's voicemail. Is that possible? I'd envision just requesting that
>> the member press "1" or something to accept the call. I currently have
>> the timeout in queues.conf set low enough so that the call will never
>> automatically roll over to that member's mobile voicemail, but I can't
>> guaranty that the staff member won't just hit "Ignore" on their phone
>> and send it directly to voicemail.
>
> You'd probably want to look at using the local channel and the followme
> application + /etc/asterisk/followme.conf
>
Full details:
1) create an entry per engineer in followme.conf
2) add each engineer to your queue as Local/[EMAIL PROTECTED]
3) create a followme context in extensions.conf
=followme.conf=
[bob]
number=>07973000123
[jim]
number=>07973000124
=queue.conf=
[support]
member => Local/[EMAIL PROTECTED]
member => Local/[EMAIL PROTECTED]
=extensons.conf=
[meetme]
exten => ._,1,MeetMe(${EXTEN})
--
Will Tatam
***
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[asterisk-users] Queue() AGI Bug ?
The docs state that the AGI is run when the caller is connected but this does not appear to be true with 1.4.21.1 What I see is 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) AGI is executed 5) agent does not answer the call before timeout, call goes to next agent 6) agent2 answers call but the AGI has already run Expected behaviour 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) agent does not answer the call before timeout, call goes to next agent 5) agent2 answers call but the AGI has already run 6) AGI is executed I need the AGI to run when the actual call is connected to an agent as my AGI is tracking which agent took the call to then fire of a jabber message to that agent giving them them the url to access the caller's account page. Currently the message is going to agent1 and agent2 who actually takes the call never sees the message -- Will Tatam *** Unite against human rights abuse in the 'war on terror' http://www.unsubscribe-me.org Amnesty International ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] [asterisk-gui] Asterisk GUI - Call Waiting
bkruse wrote: > Is this with the latest version of the gui? > > (branches/asterisknow) > (http://asteriskNOW.org/install-related) > > Tell me what revision, and paste the context of the user entry thats > having a problem. > > -bk > > > Will Tatam wrote: >> Has anyone tested disabling call waiting for a SIP extension via the GUI ? >> >> I have deselected call waiting for a user with a SNOM 360 and applied my >> changes but they still get calls waiting and are reporting that 80% of >> the time when they get the bleeping in their ear when the new call comes >> in and that it kills the current call before they get chance to respond >> in any way >> >> ___ >> --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-gui mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-gui >> > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-gui This is using asgterisk now beta 6 [8018] callwaiting=no cid_number=02380988018 context=numberplan-custom-1 [EMAIL PROTECTED] fullname=Andrew Cartlidge group= hasagent=no hasdirectory=yes hasiax=no hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=8018 secret=14731473 threewaycalling=yes zapchan= registeriax=no registersip=yes canreinvite=yes nat=no dtmfmode=rfc2833 vmsecret=1473 ___ --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] Asterisk GUI - Call Waiting
Has anyone tested disabling call waiting for a SIP extension via the GUI ? I have deselected call waiting for a user with a SNOM 360 and applied my changes but they still get calls waiting and are reporting that 80% of the time when they get the bleeping in their ear when the new call comes in and that it kills the current call before they get chance to respond in any way ___ --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] UK Male English Voices
Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham <[EMAIL PROTECTED]> who did all the hard work of recording them, and Jim Credland <[EMAIL PROTECTED]> for doing all the converting/sound work. Regards Steve The website appears to be down -- Will Tatam Email / JID [EMAIL PROTECTED] Web www.netmindz.net PGP Key www.netmindz.net/will/will_tatam.asc Registered Linux user 294695 Linux Counter http://counter.li.org See http://www.jabber.org/ to find out more about the most advanced cross platform, open source enterprise messaging solution ___ --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
Re: [Asterisk-Users] Re: [asterisk-biz] UK Male English Voices
Where are yours ? Mark Phillips wrote: Yet another set? I get about 50 downloads a week for mine. Mark On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham <[EMAIL PROTECTED]> who did all the hard work of recording them, and Jim Credland <[EMAIL PROTECTED]> for doing all the converting/sound work. Regards Steve ___ --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 -- Will Tatam Email / JID [EMAIL PROTECTED] Web www.netmindz.net PGP Key www.netmindz.net/will/will_tatam.asc Registered Linux user 294695 Linux Counter http://counter.li.org See http://www.jabber.org/ to find out more about the most advanced cross platform, open source enterprise messaging solution ___ --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
