[asterisk-users] adding a second extension

[Eric "ManxPower" Wieling]

....ast_request: No channel type registered for ''SIP'

Notice the extra ' in the message.

That is either an error in the error message or you have a an extra ' in 
your Dial line.  Something like Dial('SIP/....

I'm surprised nobody else noticed this.

Stephen Reese wrote:
> 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.
> 
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