Re: [asterisk-users] google voice calling dial plan question.
On Tue, Dec 6, 2011 at 4:05 PM, white hat whitehat...@gmail.com wrote: 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? Yes, to both of the last questions. I am using STUN and my asterisk(s) are behind a NAT device (a Netgear WND3700). My jabber.conf looks like: [general] autoregister=yes debug=yes autoprune=no auth_policy=accept [asterisk] type=client serverhost=talk.google.com ; username=xxx...@gmail.com/Talk username=xx...@gmail.com/asterisk secret=XX priority=1 port=5222 usetls=yes usesasl=yes buddy=xxx...@gmail.com status=available statusmessage=I am an Asterisk Server timeout=100 context=gtalk_incoming and, gtalk.conf looks like this: [general] context=LocalSets ; Context to dump call into bindaddr=0.0.0.0; Address to bind to allowguests=yes ; Allow calls from people not in list of peers [guest] ; special account for options on guest account disallow=all allow=ulaw context=gtalk_incoming [XX] username=xxx...@gmail.com disallow=all allow=ulaw context=gtalk_incoming connection=asterisk And, I think that just dumps incoming calls into the context that I posted previously. 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
Re: [asterisk-users] google voice calling dial plan question.
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
Re: [asterisk-users] Question about voip.ms service.
Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- 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 -- + Dave Aibel President CEO Pervasive Telecommunications, Inc. email: dai...@pervasivetelecom.com (603)367.3512 (603)367.9942 (401)862.4203 (c) 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
