Luis Antonio Prata Barbosa
2007-Sep-12 17:44 UTC
[asterisk-users] Callback for unanswered transfers...
Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm
On 9/12/07, Luis Antonio Prata Barbosa <luispratalistas at gmail.com> wrote:> Hi, > > Does anybody know if there is a way for a call goes back to transferer if > unanswered ?Yes, before Dial to transferer set some variable that have he's extension, and in your defined TRANSFER_CONTEXT, use Dial with g option. After that, check "DIALSTATUS"!="ANSWERED" and Dial back. I have it working, if those details aren't enough, or something doesn't go well, just ask. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, atis at BEST.eu.org ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? -> www.BEST.eu.org
On 9/13/07, Hoai-Anh Ngo-Vi <hoaianh at gmx.de> wrote:> Have you answered the channel?Voicemail doesn't require Answer(). It does that itself, as you usually get to voicemail after Dial(). It would be silly to require to do Answer after each Dial and then send to voicemail. Regards, Atis> > Von: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag > von Jon Weisman> I've got a strange issue here. When I make a SIP call to say my voicemail > app, I hear audio just fine. However when I dial from PSTN into my Asterisk > box, I see that its playing the voice files, but I hear nothing, then the > call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output > below. T-1 is PRI, showing normal, dchannel is up as well. Any help is > greatly appreciated. > > > > > > > > > Thanks, > > > Jon > > > > > > > > > -- Accepting call from '2125551212' to '6465551212' on channel 0/23, span 4 > -- Executing VoiceMail("Zap/95-1", "u100") in new stack > -- Playing 'vm-theperson' (language 'en') > -- Playing 'digits/1' (language 'en') > -- Playing 'digits/0' (language 'en') > -- Playing 'digits/0' (language 'en') > -- Playing 'vm-isunavail' (language 'en') > -- Playing 'vm-intro' (language 'en') > -- Channel 0/23, span 4 got hangup request, cause 34 > == Spawn extension (default, 6465551212, 1) exited non-zero on 'Zap/95-1' > -- Hungup 'Zap/95-1' > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Atis Lezdins, IT Responsible of BEST Riga, atis at BEST.eu.org ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? -> www.BEST.eu.org