<div dir="ltr">The second call its OK, so the problem it is just with the Dial(SIP/102), so try:<div><br></div><div>originate SIP/102 application Dial SIP/102</div><div><br></div><div>and </div><div><br></div><div>originate SIP/101 application Dial SIP/102</div>
<div><br></div><div>and </div><div><br></div><div>originate SIP/102 application Dial SIP/101</div><div><br></div><div><br></div><div><br></div><div><br><br><div class="gmail_quote">On Sun, Oct 19, 2008 at 11:46 PM, Stephen Reese <span dir="ltr"><<a href="mailto:rsreese@gmail.com">rsreese@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="Ih2E3d">On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez <<a href="mailto:jerdguez@gmail.com">jerdguez@gmail.com</a>> wrote:<br>
> Stephen:<br>
> Your configuration files looks fine. Try from the CLI issuing "originate<br>
> SIP/101 extension 102@default", having the 101 online, then do that with<br>
> "originate SIP/102 extension 101@default". See what happens.<br>
> If you got a CVS commit, commit again or try installing a release.<br>
> <a href="http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz" target="_blank">http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz</a> (for<br>
> download)<br>
> Regards,<br>
> Juan<br>
<br>
</div>I grabbed the latest tarball and installed it.<br>
<br>
The extension rings through to 101 and then when I answer it tries to<br>
ring through to 102 but seems to fail.<br>
<br>
ns1*CLI> originate SIP/101 extension 102@default<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [102@default:1] Dial("SIP/101-08245390",<br>
<div class="Ih2E3d">"'SIP/102',20") in new stack<br>
</div>[Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No<br>
<div class="Ih2E3d">channel type registered for ''SIP'<br>
</div>[Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full:<br>
<div class="Ih2E3d">Unable to create channel of type ''SIP' (cause 66 - Channel not<br>
implemented)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br>
</div> -- Executing [102@default:2] Hangup("SIP/101-08245390", "") in new stack<br>
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390'<br>
<br>
The extension rings through to 102 and when I answer the line it<br>
begins to ring line 101.<br>
<br>
ns1*CLI> originate SIP/102 extension 101@default<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [101@default:1] Dial("SIP/102-08249e28",<br>
"SIP/101&SIP/9046260705@vitel-outbound,30") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called 101<br>
== Using SIP RTP CoS mark 5<br>
-- Called 9046260705@vitel-outbound<br>
-- SIP/101-08244e88 is ringing<br>
-- SIP/vitel-outbound-0825d1e0 is making progress passing it to<br>
SIP/102-08249e28<br>
-- SIP/vitel-outbound-0825d1e0 is ringing<br>
-- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28<br>
-- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0<br>
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28'<br>
<br>
I'm at a loss. Thanks for your help.<br>
</blockquote></div><br><br clear="all"><br>-- <br>Juan E. Rodríguez<br>Cel. 829-886-5565<br>Work: 809-724-9227<br>
</div></div>