You could also try putting a Progress() statement between Answer and Wait. I know there is a latency issue with DAHDI calls; 5 seconds may or may not be enough for googlevoice.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of white hat Sent: Tuesday, December 06, 2011 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] google voice calling dial plan question. dwa As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped. I'm sure it's a dial plan configuration issue. Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Thanks On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel <dai...@pervasivetelecom.com> wrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat <whitehat...@gmail.com> wrote: > When a caller calls my google voice phone number, I must answer, wait and > press one to accept. Sometimes even that does not work. > > > I just need a little advice on how to write the dial plan. I still have > much to learn about asterisk, and appreciate any advice. > Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten => s,1,Answer() exten => s,n,Wait(5) exten => s,n,SendDTMF(1) exten => s,n,Dial(SIP/Ciscofficephone,10) exten => s,n,Playback(vm-nobodyavail) exten => s,n,Playback(vm-pls-try-again) same => n,Hangup() HTH, dwa dai...@pervasivetelcom.com -- _____________________________________________________________________ -- 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