Chris,<div><br></div><div>Thanks for answering my message.</div><div><br></div><div>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.</div>
<div><br></div><div>Are you suggesting I upgrade to a later version of Asterisk?</div><div><br></div><div>Thanks</div><div><br></div><div>Mitch</div><div><br></div><div>Message: 14<br>Date: Wed, 24 Oct 2012 10:20:37 -0500<br>
From: Christopher Harrington <<a href="mailto:chris@acsdi.com">chris@acsdi.com</a>><br>Subject: Re: [asterisk-users] as soon as Phone rings I'm disconnected<br> yet phone rings two more times?<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>Message-ID:<br> <CAJLBXEnd9nyyLCpAvwcD+s5jsCL8qS=cYgnUqAuS=<a href="mailto:Qay8K26CQ@mail.gmail.com">Qay8K26CQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br><br>On Tue, Oct 23, 2012 at 7:54 PM, Mitchell Johnson<br><<a href="mailto:mitch.johnson7@gmail.com">mitch.johnson7@gmail.com</a>>wrote:<br><br>><br>> One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal<br>
> adapter onto my asterisk.<br>><br><br>What version of Asterisk are you using?<br><br>[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite:<br>> Call from '' (<a href="http://172.16.200.1:65451/" target="_blank">172.16.200.1:65451</a>) to extension '5000' rejected because<br>
> extension not found in context 'default'.<br>><br><br>Did you mean to include this notice in your email? It indicates a dialplan<br>problem.<br><br><br>> -- Executing [5000@pstn-incoming:1] Dial("SIP/172.16.200.1-00000006",<br>
> "SIP/5000,20|p") in new stack<br>><br><br>The pipe has been deprecated in more recent versions of Asterisk, make sure<br>this isn't related to your issue.<br><br>--<br>-Chris Harrington<br>ACSDi Office: <a href="tel:763.559.5800" value="+17635595800">763.559.5800</a><br>
Mobile Phone: <a href="tel:612.326.4248" value="+16123264248">612.326.4248</a><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20121024/fef3e983/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20121024/fef3e983/attachment-0001.htm</a>><br>
<br>------------------------------<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
<br>AstriCon 2010 - October 26-28 Washington, DC<br>Register Now: <a href="http://www.astricon.net/" target="_blank">http://www.astricon.net/</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br>End of asterisk-users Digest, Vol 99, Issue 37<br>**********************************************<br>
<div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Message: 12<br>
Date: Tue, 23 Oct 2012 20:54:50 -0400<br>
From: Mitchell Johnson <<a href="mailto:mitch.johnson7@gmail.com">mitch.johnson7@gmail.com</a>><br>
Subject: [asterisk-users] as soon as Phone rings I'm disconnected yet<br>
phone rings two more times?<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Message-ID: <<a href="mailto:C9E439FA-355F-4EAF-A42A-652C2C84412C@gmail.com">C9E439FA-355F-4EAF-A42A-652C2C84412C@gmail.com</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
<br>
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:<br>
<br>
voice-port 0/2/0<br>
connection plar opx 5000<br>
caller-id enable<br>
<br>
dial-peer voice 200 voip<br>
destination-pattern 5...<br>
session protocol sipv2<br>
session target sip-server<br>
codec g711ulaw<br>
!<br>
sip-ua<br>
sip-server ipv4:172.16.200.212 <------ Asterisk server<br>
<br>
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.<br>
<br>
The following output is what I see on the Asterisk console:<br>
<br>
asterisk*CLI><br>
== Using SIP RTP CoS mark 5<br>
[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: Call from '' (<a href="http://172.16.200.1:65451" target="_blank">172.16.200.1:65451</a>) to extension '5000' rejected because extension not found in context 'default'.<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [5000@pstn-incoming:1] Dial("SIP/172.16.200.1-00000006", "SIP/5000,20|p") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/5000<br>
-- SIP/5000-00000007 is ringing<br>
== Spawn extension (pstn-incoming, 5000, 1) exited non-zero on 'SIP/172.16.200.1-00000006'<br>
<br>
The 172.16.200.1 is my router.<br>
<br>
sip.conf excerpt:<br>
<br>
[5000]<br>
type=friend<br>
context=phones<br>
host=dynamic<br>
disallow=all<br>
allow=ulaw<br>
secret=cisco123<br>
mailbox=5000@phones<br>
<br>
[172.16.200.1]<br>
context=pstn-incoming<br>
type=friend<br>
host=172.16.200.1<br>
dtmfmode=rfc2833<br>
disallow=all<br>
allow=ulaw<br>
<br>
[phones]<br>
exten => 5000,1,Dial(SIP/${EXTEN},20|p)<br>
exten => 5000,n,Hangup<br>
<br>
[pstn-incoming]<br>
include=phones<br>
<br>
Any help would be greatly appreciated,<br>
<br>
Thanks,<br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20121023/6765c48b/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20121023/6765c48b/attachment-0001.htm</a>><br>
<br>
------------------------------<br>
<br><br></blockquote></div></div>