mitch Johnson
2012-Oct-26 01:09 UTC
[asterisk-users] asterisk-users Digest, Vol 99, Issue 37
Chris, Thanks for answering my message. I'm currently using version 10.5.1. I included the error message on the dial plan to show what errors I was displaying. The call goes through after that error message is displayed. As soon as I hear the phone ring, it drops my call on the calling phone yet the called phone rings two more times before being dropped, also. Are you suggesting I upgrade to a later version of Asterisk? Thanks Mitch Message: 14 Date: Wed, 24 Oct 2012 10:20:37 -0500 From: Christopher Harrington <chris at acsdi.com> Subject: Re: [asterisk-users] as soon as Phone rings I'm disconnected yet phone rings two more times? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <CAJLBXEnd9nyyLCpAvwcD+s5jsCL8qS=cYgnUqAuS=Qay8K26CQ at mail.gmail.com> Content-Type: text/plain; charset="utf-8" On Tue, Oct 23, 2012 at 7:54 PM, Mitchell Johnson <mitch.johnson7 at gmail.com>wrote:> > One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal > adapter onto my asterisk. >What version of Asterisk are you using? [Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite:> Call from '' (172.16.200.1:65451) to extension '5000' rejected because > extension not found in context 'default'. >Did you mean to include this notice in your email? It indicates a dialplan problem.> -- Executing [5000 at pstn-incoming:1] Dial("SIP/172.16.200.1-00000006", > "SIP/5000,20|p") in new stack >The pipe has been deprecated in more recent versions of Asterisk, make sure this isn't related to your issue. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -------------- next part -------------- An HTML attachment was scrubbed... URL: < http://lists.digium.com/pipermail/asterisk-users/attachments/20121024/fef3e983/attachment-0001.htm>------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 99, Issue 37 **********************************************> > Message: 12 > Date: Tue, 23 Oct 2012 20:54:50 -0400 > From: Mitchell Johnson <mitch.johnson7 at gmail.com> > Subject: [asterisk-users] as soon as Phone rings I'm disconnected yet > phone rings two more times? > To: asterisk-users at lists.digium.com > Message-ID: <C9E439FA-355F-4EAF-A42A-652C2C84412C at gmail.com> > Content-Type: text/plain; charset="us-ascii" > > > One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal > adapter onto my asterisk. I have the 8x8 box connected to the Internet, > and the phone line connected to an fxo port on a Cisco router: > > voice-port 0/2/0 > connection plar opx 5000 > caller-id enable > > dial-peer voice 200 voip > destination-pattern 5... > session protocol sipv2 > session target sip-server > codec g711ulaw > ! > sip-ua > sip-server ipv4:172.16.200.212 <------ Asterisk server > > When I make a call from the PSTN to the 8x8 box, it does send ring back to > the asterisk server and the Digium phone does ring. However, as soon as > the phone rings the call disconnects yet the actual phone, extension 5000, > rings two times before it hangs up, also. > > The following output is what I see on the Asterisk console: > > asterisk*CLI> > == Using SIP RTP CoS mark 5 > [Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: > Call from '' (172.16.200.1:65451) to extension '5000' rejected because > extension not found in context 'default'. > == Using SIP RTP CoS mark 5 > -- Executing [5000 at pstn-incoming:1] Dial("SIP/172.16.200.1-00000006", > "SIP/5000,20|p") in new stack > == Using SIP RTP CoS mark 5 > -- Called SIP/5000 > -- SIP/5000-00000007 is ringing > == Spawn extension (pstn-incoming, 5000, 1) exited non-zero on > 'SIP/172.16.200.1-00000006' > > The 172.16.200.1 is my router. > > sip.conf excerpt: > > [5000] > type=friend > context=phones > host=dynamic > disallow=all > allow=ulaw > secret=cisco123 > mailbox=5000 at phones > > [172.16.200.1] > context=pstn-incoming > type=friend > host=172.16.200.1 > dtmfmode=rfc2833 > disallow=all > allow=ulaw > > [phones] > exten => 5000,1,Dial(SIP/${EXTEN},20|p) > exten => 5000,n,Hangup > > [pstn-incoming] > include=phones > > Any help would be greatly appreciated, > > Thanks, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20121023/6765c48b/attachment-0001.htm > > > > ------------------------------ > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121025/c1e75f3e/attachment.htm>