[asterisk-users] adding a second extension
Stephen Reese
rsreese at gmail.com
Sun Oct 19 22:46:38 CDT 2008
On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez <jerdguez at gmail.com> wrote:
> Stephen:
> Your configuration files looks fine. Try from the CLI issuing "originate
> SIP/101 extension 102 at default", having the 101 online, then do that with
> "originate SIP/102 extension 101 at default". See what happens.
> If you got a CVS commit, commit again or try installing a release.
> http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for
> download)
> Regards,
> Juan
I grabbed the latest tarball and installed it.
The extension rings through to 101 and then when I answer it tries to
ring through to 102 but seems to fail.
ns1*CLI> originate SIP/101 extension 102 at default
== Using SIP RTP CoS mark 5
-- Executing [102 at default:1] Dial("SIP/101-08245390",
"'SIP/102',20") in new stack
[Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [102 at default:2] Hangup("SIP/101-08245390", "") in new stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390'
The extension rings through to 102 and when I answer the line it
begins to ring line 101.
ns1*CLI> originate SIP/102 extension 101 at default
== Using SIP RTP CoS mark 5
-- Executing [101 at default:1] Dial("SIP/102-08249e28",
"SIP/101&SIP/9046260705 at vitel-outbound,30") in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called 9046260705 at vitel-outbound
-- SIP/101-08244e88 is ringing
-- SIP/vitel-outbound-0825d1e0 is making progress passing it to
SIP/102-08249e28
-- SIP/vitel-outbound-0825d1e0 is ringing
-- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28
-- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28'
I'm at a loss. Thanks for your help.
More information about the asterisk-users
mailing list