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 <zisha...@gmail.com> Subject: Re: [asterisk-users] Local channel usage To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <aanlktilo2hzyq4jp7rb_iguzaq-n2chxnhq96grg0...@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.ge...@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 <ha...@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