Re: [asterisk-users] Call Return
Hi Aj
Can you perhaps show me an example as to how you would do it as I have tried
setting it very early but still doesn’t work
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net
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-Original Message-
From: A J Stiles [mailto:asterisk_l...@earthshod.co.uk]
Sent: Thursday, July 9, 2015 10:03 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Return
On Wednesday 08 Jul 2015, Andrew Colin wrote:
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is
not answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for
example someone calls in to the switchboard and they transfer it then
it tries to return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in
this case 200) and declare it as a variable?
You need to set a variable quite early in your extension logic, using a Set
command;
Set(dialled=${EXTEN})
and then later you can retrieve it as ${dialled} . This variable will
persist across context jumps, even although ${EXTEN} may have changed.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
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Re: [asterisk-users] Call Return
On Wednesday 08 Jul 2015, Andrew Colin wrote:
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?
You need to set a variable quite early in your extension logic, using a Set
command;
Set(dialled=${EXTEN})
and then later you can retrieve it as ${dialled} . This variable will persist
across context jumps, even although ${EXTEN} may have changed.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- 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 Return
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?
--
_
-- 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 Return
does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove ___ --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 Return
I think you can do it in extensions.conf. I would use a special macro only for the operator, where the call would be directed back to the operator instead of voicemail, if the transferee is busy or unavailable. Sounds simple to me. - Original Message - From: Paradise Dove [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 02, 2005 9:52 PM Subject: [Asterisk-Users] Call Return does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove ___ --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 return?
For me this worked straight out of the box with [EMAIL PROTECTED] 0.3 Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Polk Sent: Sunday, 23 January 2005 4:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call return? Hi: Can any one point me in the rite direction on this? I am using asterisk at home for learning purposes. I am trying to get the triditional *69 working. Has there been any success in getting it to announce the number and get it to give you the option to call back? Chris - Original Message - From: Diego Ventrice [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 8:03 AM Subject: Re: [Asterisk-Users] softswitch dilemma Thanks for answering Chad, Actually, I just want to Switch traffic between wholesale providers (my customers) which actually terminate traffic (or not, some of them have just controllers-softswitches like the one Im willing to set up) collect CDRs and bill them =) I have no gateways of my own (of any kind) so Im not originating nor terminating calls, just switching traffic is my goal, all this people use h.323 of course. Any advice would be appreciated. Thanks for your help D. Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST) From: Chad Whitten [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] softswitch dilemma To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 are you looking to do actual pstn to voip termination? if so, then you are gonna need ss7, cama and imt trunks - things which asterisk doesnt necessarily support. now if you just want to buy pri/t1 from the local telco and sell voip services off an asterisk server that gets back to the pstn over these pri/t1's, then yes, asterisk can do this. Diego Ventrice said: Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that. An asterisk-ser or an asterisk-vocal combination may be the answer ? Thanks in advance for any help. Diego -- Chad Whitten ___ 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 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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
[Asterisk-Users] call return?
Hi: Can any one point me in the rite direction on this? I am using asterisk at home for learning purposes. I am trying to get the triditional *69 working. Has there been any success in getting it to announce the number and get it to give you the option to call back? Chris - Original Message - From: Diego Ventrice [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 8:03 AM Subject: Re: [Asterisk-Users] softswitch dilemma Thanks for answering Chad, Actually, I just want to Switch traffic between wholesale providers (my customers) which actually terminate traffic (or not, some of them have just controllers-softswitches like the one Im willing to set up) collect CDRs and bill them =) I have no gateways of my own (of any kind) so Im not originating nor terminating calls, just switching traffic is my goal, all this people use h.323 of course. Any advice would be appreciated. Thanks for your help D. Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST) From: Chad Whitten [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] softswitch dilemma To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 are you looking to do actual pstn to voip termination? if so, then you are gonna need ss7, cama and imt trunks - things which asterisk doesnt necessarily support. now if you just want to buy pri/t1 from the local telco and sell voip services off an asterisk server that gets back to the pstn over these pri/t1's, then yes, asterisk can do this. Diego Ventrice said: Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that. An asterisk-ser or an asterisk-vocal combination may be the answer ? Thanks in advance for any help. Diego -- Chad Whitten ___ 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 ___ 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
