Re: [asterisk-users] Dialing out and recording
Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8
/H
Den 2013-01-02 22:25 skrev Danny Nicholas :
>1.6.2 is a "deader soldier" than 1.4.X.
>
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
>Westerberg
>Sent: Wednesday, January 02, 2013 3:20 PM
>To: asterisk-users@lists.digium.com
>Subject: Re: [asterisk-users] Dialing out and recording
>
>#2 works for me on Asterisk 1.8.12 when setting the header like this:
>
>exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
>
>I haven't been able to make it work on 1.6 yet though, has anyone else?
>
>
>/Henrik
>
>
>>
>>
>>
>>
>>
>>From: asterisk-users-boun...@lists.digium.com
>>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
>>Sent: Wednesday, January 02, 2013 9:32 AM
>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>Subject: Re: [asterisk-users] Dialing out and recording
>>
>>
>>
>>I have the same requirement, but it's important that the caller ID
>>information from the original caller is presented to the destination
>>and we announce the call before the "transfer" is complete. The carrier
>>requires a diversion header if the ANI is not one of "our" DIDs. Does
>>someone have experience with this working?
>>
>>--
>>
>>Two suggestions for you, Don. #1 if the Dial is "Private" the
>>"announcement" is taken care of. #2 I'm supposing that you could do a
>>"SIP Header" command before the Dial to resolve the diversion header
>>issue.
>>
>>-- next part -- An HTML attachment was
>>scrubbed...
>>URL:
>><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/
>>459
>>43b1f/attachment-0001.htm>
>>
>>--
>
>
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Re: [asterisk-users] Dialing out and recording
1.6.2 is a "deader soldier" than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
>
>
>
>
>
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
>Sent: Wednesday, January 02, 2013 9:32 AM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Dialing out and recording
>
>
>
>I have the same requirement, but it's important that the caller ID
>information from the original caller is presented to the destination
>and we announce the call before the "transfer" is complete. The carrier
>requires a diversion header if the ANI is not one of "our" DIDs. Does
>someone have experience with this working?
>
>--
>
>Two suggestions for you, Don. #1 if the Dial is "Private" the
>"announcement" is taken care of. #2 I'm supposing that you could do a
>"SIP Header" command before the Dial to resolve the diversion header issue.
>
>-- next part -- An HTML attachment was
>scrubbed...
>URL:
><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/
>459
>43b1f/attachment-0001.htm>
>
>--
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Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
>
>
>
>
>
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
>Sent: Wednesday, January 02, 2013 9:32 AM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Dialing out and recording
>
>
>
>I have the same requirement, but it's important that the caller ID
>information from the original caller is presented to the destination and
>we
>announce the call before the "transfer" is complete. The carrier requires
>a
>diversion header if the ANI is not one of "our" DIDs. Does someone have
>experience with this working?
>
>--
>
>Two suggestions for you, Don. #1 if the Dial is "Private" the
>"announcement" is taken care of. #2 I'm supposing that you could do a "SIP
>Header" command before the Dial to resolve the diversion header issue.
>
>-- next part --
>An HTML attachment was scrubbed...
>URL:
><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/459
>43b1f/attachment-0001.htm>
>
>--
--
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Re: [asterisk-users] Dialing out and recording
Thanks Danny I will try this.
/Henrik
>
>Message: 12
>Date: Wed, 2 Jan 2013 08:17:59 -0600
>From: "Danny Nicholas"
>Subject: Re: [asterisk-users] Dialing out and recording
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
>Message-ID: <001501cde8f3$f7d2b290$e77817b0$@debsinc.com>
>Content-Type: text/plain; charset="us-ascii"
>
>Put the AGI call in a macro context and add M(macro) to your Dial string.
>
>
>
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
>Westerberg
>Sent: Wednesday, January 02, 2013 8:02 AM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] Dialing out and recording
>
>
>
>Hi,
>
>
>
>I am using asterisk via AGI and want to be able to record a call.
>
>The scenario is:
>
>1. A call comes in
>2. The call is redirected to a mobile number via a local extension and
>ChannelRedirect
>3. The local extension looks like something this:
>
>exten => _X.,1,Dial(SIP/${EXTEN},60,.)
>
>exten => _X.,n,Agi(agi://localhost/aj.agi?action=)
>
>
>
>I have looked through all arguments of Dial but haven't found any way to
>continue having a connected call between the caller and the callee and
>have
>AGI control of it. Is there a way to do this or do I have to use G() and
>connect the both ends to AGI separately and then bridging them before
>recording the call?
>
>
>
>Thanks for help.
>
>
>
>Regards,
>
>
>
>Henrik
>
>-- next part --
>An HTML attachment was scrubbed...
>URL:
><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/ce6
>b7c57/attachment-0001.htm>
>
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Re: [asterisk-users] Dialing out and recording
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,.)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the "transfer" is complete. The carrier requires a
diversion header if the ANI is not one of "our" DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is "Private" the
"announcement" is taken care of. #2 I'm supposing that you could do a "SIP
Header" command before the Dial to resolve the diversion header issue.
--
_
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Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the "transfer" is complete. The carrier requires a
diversion header if the ANI is not one of "our" DIDs. Does someone have
experience with this working?
(Top-posting 'cause the last guy did)
--Don
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro) to your Dial string.
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,.)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Dialing out and recording
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,.)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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[asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,…)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=……..)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have AGI
control of it. Is there a way to do this or do I have to use G() and connect
the both ends to AGI separately and then bridging them before recording the
call?
Thanks for help.
Regards,
Henrik
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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