[asterisk-users] Call Back on Busy

2011-01-10 Thread Ron

Hi All,

One of our user asked the question, when she tries to call another local 
extension but the other end is engaged she will keep on trying until she 
finally can get thru. So she asked would it be possible to request for 
an auto-callback from the user she's trying to call to once it's not 
engaged anymore. is this possible on asterisk? what is that feature 
called? i am using asterisk 1.4 with freepbx. Thanks in advance.


Regards
Ron

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread John Novack

That function in the telephony world is called camp-on

Can't say for sure if Asterisk can do that, not which version, nor freepbx

John Novack

Ron wrote:

Hi All,

One of our user asked the question, when she tries to call another 
local extension but the other end is engaged she will keep on trying 
until she finally can get thru. So she asked would it be possible to 
request for an auto-callback from the user she's trying to call to 
once it's not engaged anymore. is this possible on asterisk? what is 
that feature called? i am using asterisk 1.4 with freepbx. Thanks in 
advance.


Regards
Ron

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



--

Dog is my Co-pilot


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Jim Dickenson
It should not be too hard to write some dialplan code that detects the busy, 
plays a sound file asking if you want to camp-on to the called device, read an 
answer and loop around checking device status and placing a call when both the 
calling device and called device are free.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 10, 2011, at 8:39 AM, John Novack wrote:

 That function in the telephony world is called camp-on
 
 Can't say for sure if Asterisk can do that, not which version, nor freepbx
 
 John Novack
 
 Ron wrote:
 Hi All,
 
 One of our user asked the question, when she tries to call another local 
 extension but the other end is engaged she will keep on trying until she 
 finally can get thru. So she asked would it be possible to request for an 
 auto-callback from the user she's trying to call to once it's not engaged 
 anymore. is this possible on asterisk? what is that feature called? i am 
 using asterisk 1.4 with freepbx. Thanks in advance.
 
 Regards
 Ron
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 -- 
 
 Dog is my Co-pilot
 
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello
 
 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Paul Belanger
On 11-01-10 09:57 AM, Ron wrote:
 One of our user asked the question, when she tries to call another local
 extension but the other end is engaged she will keep on trying until she
 finally can get thru. So she asked would it be possible to request for
 an auto-callback from the user she's trying to call to once it's not
 engaged anymore. is this possible on asterisk? what is that feature
 called? i am using asterisk 1.4 with freepbx. Thanks in advance.
 
Asterisk 1.8 - Call Completion Supplementary Services (CCSS)[1]

[1] -
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Ron
thanks for all the reply. now that i know what it's called should be 
easy to find something on the net.


btw, the URL below did not load anything on my side...it seems like it's 
connected somewhere but just downloading slow, but thanks for it anyway.


regards
Ron

On 1/11/2011 1:20 AM, Paul Belanger wrote:

On 11-01-10 09:57 AM, Ron wrote:

One of our user asked the question, when she tries to call another local
extension but the other end is engaged she will keep on trying until she
finally can get thru. So she asked would it be possible to request for
an auto-callback from the user she's trying to call to once it's not
engaged anymore. is this possible on asterisk? what is that feature
called? i am using asterisk 1.4 with freepbx. Thanks in advance.


Asterisk 1.8 - Call Completion Supplementary Services (CCSS)[1]

[1] -
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Back On Busy?

2005-09-26 Thread Sherwood McGowan



I know it's been 
touched on before, but no answers have been found to the best of my knowledge. 
I'm using a SIP only setup, with a sip provider giving PSTN and would like to 
see if anyone has an idea for creating redial busy using ${DIALSTATUS} and 
possibly MeetMe?

I figure something 
like this, but want to get feedback

1. Get callers last 
dialed number, if international number, do not allow.
2. Playback a 
stuttertone to caller
3. Disconnect 
caller
4. Ring intended 
party check dial status. If busy, wait120 seconds and try again (do this 
for a total of 15 minutes)
5. If it's picked 
up, playback an announcement to the party and put them in a meetme 
conference
6. Ring the original 
caller and bridge them to the meetme conference. 

___
--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

RE: [Asterisk-Users] Call Back On Busy?

2005-09-26 Thread Damon Estep








This may not apply to your situation, but
many ATAs and SIP phones have this feature built in to the device.



We use Linksys/Sipura and auto redial and
last call return work without any special setup.













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Monday, September 26, 2005
7:45 AM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call
Back On Busy?







I know it's been touched on before, but no answers have been
found to the best of my knowledge. I'm using a SIP only setup, with a sip
provider giving PSTN and would like to see if anyone has an idea for creating
redial busy using ${DIALSTATUS} and possibly MeetMe?











I figure something like this, but want to get feedback











1. Get callers last dialed number, if international number,
do not allow.





2. Playback a stuttertone to caller





3. Disconnect caller





4. Ring intended party check dial status. If busy,
wait120 seconds and try again (do this for a total of 15 minutes)





5. If it's picked up, playback an announcement to the party
and put them in a meetme conference





6. Ring the original caller and bridge them to the meetme
conference. 
















___
--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

RE: [Asterisk-Users] Call Back On Busy?

2005-09-26 Thread Sherwood McGowan



Thank you, I do appreciate that many ATAs have redial on 
busy, but I've been given the charge of figuring out how one would do it in 
Asterisk.

Don't ask me why


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Damon 
  EstepSent: Monday, September 26, 2005 10:15 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Call Back On Busy?
  
  
  This may not apply to 
  your situation, but many ATAs and SIP phones have this feature built in to the 
  device.
  
  We use Linksys/Sipura 
  and auto redial and last call return work without any special 
  setup.
  
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45 
  AMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] Call Back On 
  Busy?
  
  
  I know it's been touched on 
  before, but no answers have been found to the best of my knowledge. I'm using 
  a SIP only setup, with a sip provider giving PSTN and would like to see if 
  anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly 
  MeetMe?
  
  
  
  I figure something like this, but 
  want to get feedback
  
  
  
  1. Get callers last dialed number, 
  if international number, do not allow.
  
  2. Playback a stuttertone to 
  caller
  
  3. Disconnect 
  caller
  
  4. Ring intended party check dial 
  status. If busy, wait120 seconds and try again (do this for a total of 
  15 minutes)
  
  5. If it's picked up, playback an 
  announcement to the party and put them in a meetme 
  conference
  
  6. Ring the original caller and 
  bridge them to the meetme conference. 
  
  
___
--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

RE: [Asterisk-Users] Call Back On Busy?

2005-09-26 Thread Sherwood McGowan



Anyone else out there have some thoughts? The customer 
wants to be able to control what can be redialed on busy, such as no 
international. I'm having my doubts as to whether or not this can be done. My 
idea seems like it would work, but after the customer hangs up, wouldn't the 
context stop processing?

Thanks,
SKM

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Damon 
  EstepSent: Monday, September 26, 2005 10:15 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Call Back On Busy?
  
  
  This may not apply to 
  your situation, but many ATAs and SIP phones have this feature built in to the 
  device.
  
  We use Linksys/Sipura 
  and auto redial and last call return work without any special 
  setup.
  
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45 
  AMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] Call Back On 
  Busy?
  
  
  I know it's been touched on 
  before, but no answers have been found to the best of my knowledge. I'm using 
  a SIP only setup, with a sip provider giving PSTN and would like to see if 
  anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly 
  MeetMe?
  
  
  
  I figure something like this, but 
  want to get feedback
  
  
  
  1. Get callers last dialed number, 
  if international number, do not allow.
  
  2. Playback a stuttertone to 
  caller
  
  3. Disconnect 
  caller
  
  4. Ring intended party check dial 
  status. If busy, wait120 seconds and try again (do this for a total of 
  15 minutes)
  
  5. If it's picked up, playback an 
  announcement to the party and put them in a meetme 
  conference
  
  6. Ring the original caller and 
  bridge them to the meetme conference. 
  
  
___
--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

Re: [Asterisk-Users] Call Back On Busy?

2005-09-26 Thread BJ Weschke
Is there a functional reason why you'd use MeetMe here? I think probably the easiest way to accomplish this is to use an DeadAGI script which can be invoked via the 'h' extension in the context that would then perform the functionality you're looking for and if they get through it should just bridge the original caller back in. 

On 9/26/05, Sherwood McGowan [EMAIL PROTECTED] wrote:

Anyone else out there have some thoughts? The customer wants to be able to control what can be redialed on busy, such as no international. I'm having my doubts as to whether or not this can be done. My idea seems like it would work, but after the customer hangs up, wouldn't the context stop processing?


Thanks,
SKM



From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Damon EstepSent: Monday, September 26, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Call Back On Busy?



This may not apply to your situation, but many ATAs and SIP phones have this feature built in to the device.


We use Linksys/Sipura and auto redial and last call return work without any special setup.






From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Back On Busy?


I know it's been touched on before, but no answers have been found to the best of my knowledge. I'm using a SIP only setup, with a sip provider giving PSTN and would like to see if anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly MeetMe?




I figure something like this, but want to get feedback



1. Get callers last dialed number, if international number, do not allow.

2. Playback a stuttertone to caller

3. Disconnect caller

4. Ring intended party check dial status. If busy, wait120 seconds and try again (do this for a total of 15 minutes)


5. If it's picked up, playback an announcement to the party and put them in a meetme conference

6. Ring the original caller and bridge them to the meetme conference. 

___--Bandwidth and Colocation sponsored by 
Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--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

RE: [Asterisk-Users] Call Back On Busy?

2005-09-26 Thread Sherwood McGowan



Not a bad idea, thank you for that. I'll look into 
it


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of BJ 
  WeschkeSent: Monday, September 26, 2005 2:37 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Call Back On Busy?
  Is there a functional reason why you'd use MeetMe here? I 
  think probably the easiest way to accomplish this is to use an DeadAGI script 
  which can be invoked via the 'h' extension in the context that would then 
  perform the functionality you're looking for and if they get through it should 
  just bridge the original caller back in. 
  On 9/26/05, Sherwood 
  McGowan [EMAIL PROTECTED] 
  wrote: 
  
Anyone 
else out there have some thoughts? The customer wants to be able to control 
what can be redialed on busy, such as no international. I'm having my doubts 
as to whether or not this can be done. My idea seems like it would work, but 
after the customer hangs up, wouldn't the context stop processing? 


Thanks,
SKM

  
  
  From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of 
  Damon EstepSent: Monday, September 26, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [Asterisk-Users] Call Back On 
  Busy?
  
  
  
  This may not 
  apply to your situation, but many ATAs and SIP phones have this feature 
  built in to the device. 
  
  We use 
  Linksys/Sipura and auto redial and last call return work without any 
  special setup.
  
  
  
  
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Sherwood 
  McGowanSent: Monday, 
  September 26, 2005 7:45 AMTo: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' Subject: [Asterisk-Users] Call Back 
  On Busy?
  
  
  I know it's been touched on 
  before, but no answers have been found to the best of my knowledge. I'm 
  using a SIP only setup, with a sip provider giving PSTN and would like to 
  see if anyone has an idea for creating redial busy using ${DIALSTATUS} and 
  possibly MeetMe? 
  
  
  
  I figure something like this, 
  but want to get feedback
  
  
  
  1. Get callers last dialed 
  number, if international number, do not allow.
  
  2. Playback a stuttertone to 
  caller
  
  3. Disconnect 
  caller
  
  4. Ring intended party check 
  dial status. If busy, wait120 seconds and try again (do this for a 
  total of 15 minutes)
  
  5. If it's picked up, playback 
  an announcement to the party and put them in a meetme 
  conference
  
  6. Ring the original caller 
  and bridge them to the meetme conference. 
  
  ___--Bandwidth 
and Colocation sponsored by Easynews.com 
--Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users 
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 

___
--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