Hi All,
I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to
called party, calling party, etc.). I?m trying to use the local channel for this
purpose but so far with no success. I?m using 1.6.1.18 and this is my
extensions.conf:
[Internal]
exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number
exten => _22,2,Noop(After Hangup)
[CW]
exten => _x.,1,Dial(SIP/307)
exten => _x.,2,Noop(After Hangup)
The call never reaches neither of the Noop applications. Consol:
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [22 at Internal:1] Dial("SIP/309-000000a5",
"Local/22 at CW/n") in new stack
-- Called 22 at CW/n
-- Executing [22 at CW:1] Dial("Local/22 at CW-af6f;2",
"SIP/307") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 307
-- SIP/307-000000a6 is ringing
-- Local/22 at CW-af6f;1 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 answered Local/22 at CW-af6f;2
-- Local/22 at CW-af6f;1 answered SIP/309-000000a5
== Spawn extension (CW, 22, 1) exited non-zero on 'Local/22 at
CW-af6f;2'
== Spawn extension (Internal, 22, 1) exited non-zero on
'SIP/309-000000a5'
If I use the ?g? option in my Dial() both Noop will be run only if the called
party hang-up first. I need a simple continuation in the dial plan regardless of
who disconnected the call.
Thanks in advance
Harel
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Hi, After a Dial, the call is hung up. It doesn't carry on with dialplan unless you specify the appropriate dial option. Check wiki voip-info for cmd Dial, I think the option is "g" 2010/6/22 Harel Cohen <harel at easycall.gi>> Hi All, > > I?m trying to do ?things? after my Dial application terminates (e.g. play > IVR to called party, calling party, etc.). I?m trying to use the local > channel for this purpose but so far with no success. I?m using 1.6.1.18 and > this is my extensions.conf: > > > > [Internal] > > exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number > > exten => _22,2,Noop(After Hangup) > > > > [CW] > > exten => _x.,1,Dial(SIP/307) > > exten => _x.,2,Noop(After Hangup) > > > > The call never reaches neither of the Noop applications. Consol: > > == Using SIP RTP CoS mark 5 > > == Using UDPTL CoS mark 5 > > -- Executing [22 at Internal:1] Dial("SIP/309-000000a5", "Local/22 at CW/n") > in new stack > > -- Called 22 at CW/n > > -- Executing [22 at CW:1] Dial("Local/22 at CW-af6f;2", "SIP/307") in new > stack > > == Using SIP RTP CoS mark 5 > > == Using UDPTL CoS mark 5 > > -- Called 307 > > -- SIP/307-000000a6 is ringing > > -- Local/22 at CW-af6f;1 is ringing > > -- SIP/307-000000a6 is ringing > > -- SIP/307-000000a6 is ringing > > -- SIP/307-000000a6 is ringing > > -- SIP/307-000000a6 answered Local/22 at CW-af6f;2 > > -- Local/22 at CW-af6f;1 answered SIP/309-000000a5 > > == Spawn extension (CW, 22, 1) exited non-zero on 'Local/22 at CW-af6f;2' > > == Spawn extension (Internal, 22, 1) exited non-zero on > 'SIP/309-000000a5' > > If I use the ?g? option in my Dial() both Noop will be run only if the > called party hang-up first. I need a simple continuation in the dial plan > regardless of who disconnected the call. > > Thanks in advance > > Harel > > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100622/66a7e51a/attachment-0001.htm
'g' option continues the dial plan after the call has been answered, not after it is hung up. Depending upon what you are trying to do, first try to use h extension, i.e. in the example you gave, you should replace '_22,2' with 'h,1'. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-22 6:23 AM, "Tiago Geada" <tiago.geada at gmail.com> wrote: Hi, After a Dial, the call is hung up. It doesn't carry on with dialplan unless you specify the appropriate dial option. Check wiki voip-info for cmd Dial, I think the option is "g" 2010/6/22 Harel Cohen <harel at easycall.gi>> > > > Hi All, > > > > I?m trying to do ?things? after my Dial application terminates (e.g. play > IVR to cal... > -- > _____________________________________________________________________ > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100622/4097fe3c/attachment.htm
Zeeshan: 1. g option continues the dial plan after the called party hangup, and only the called party. See the manual or check for yourself... 2. h extension is no good for me because the voice path is already closed at this point therefore I cannot play IVR (Im getting Warnings like: file.c:750 ast_readaudio_callback: Failed to write frame). Tiago: There is no Dial() option to simply continue dial-plan after Dial(). See above regarding g option. Can anyone think of a way to play IVR after conversation initiated by Dial() terminates? Harel ------------------------------ Message: 9 Date: Tue, 22 Jun 2010 07:27:42 -0400 From: Zeeshan Zakaria <zishanov at gmail.com> Subject: Re: [asterisk-users] Local channel usage To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTilO2Hzyq4Jp7Rb_igUzaQ-N2chxNhq96Grg0P-u at mail.gmail.com> Content-Type: text/plain; charset="windows-1252" 'g' option continues the dial plan after the call has been answered, not after it is hung up. Depending upon what you are trying to do, first try to use h extension, i.e. in the example you gave, you should replace '_22,2' with 'h,1'. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-22 6:23 AM, "Tiago Geada" <tiago.geada at gmail.com> wrote: Hi, After a Dial, the call is hung up. It doesn't carry on with dialplan unless you specify the appropriate dial option. Check wiki voip-info for cmd Dial, I think the option is "g" 2010/6/22 Harel Cohen <harel at easycall.gi>> > > > Hi All, > > > > I?m trying to do ?things? after my Dial application terminates (e.g. play > IVR to cal... > -- > _____________________________________________________________________ > -- 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 >
I'll sure look at the 'g' option tomorrow when back at work, as I am using it on a production system. MeetMe is a good way to accomplish this, I've done it several times though a little tricky, but works great. Another idea could be to use option M^ and send the call to a macro and write a macro to handle the call after it is hung up. There is also an option 'b' which opens up an agi script. I once used it but gave up because if I remember correctly it ran the agi in parallel with the dial command. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-22 1:08 PM, "Philipp von Klitzing" < klitzing at pool.informatik.rwth-aachen.de> wrote: Hi!> Can anyone think of a way to play IVR after conversation initiated by > Dial() terminates?You will most probably have to prevent the hangup to happen in the first place: You could, for example, join the two callers by the help of a dynamic MeetMe room, and then take action when the other parties leaves, i.e. kick the remaining user out of the room and continue in the dialplan. Here's an example for "Voicemail live" that uses such a technique: http://www.voip-info.org/wiki/view/Asterisk+tips+voicemail+live Another way *might* be to involve a local channel for the calling party with the /n option to prevent it from optimizing themselves away: For example: The caller's SIP channel hangs up, but the local channel that it is connected with then continues in the dialplan? Not sure if there is a way to make this work - could be that you need to twist things badly so that also the caller is in fact a callee to the local channel... Finally: Put a SIP proxy in between that catches the hangup and then takes action like a redirect (transfer). Philipp -- _____________________________________________________________________ -- Bandwidth and Colocation Pr... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100622/bdb763ff/attachment.htm