[Freeswitch-users] SIP gateway - can't locate outgoing channel type

[Richard Weremiuk]

Anthony

 

Thanks for the response. It was exactly that.  "transfer" works fine.  I did
get the code from the book yes, and of course didn't realise there was a
known  error.  I'll have a look to see if there a list of other known book
issues on the wiki anywhere to avoid making the same mistake.

 

That's really helped.  Pulled my hair out all day with that.


Richard

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 12 October 2012 18:22
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP gateway - can't locate outgoing channel
type

 

Dialplan: sofia/external/07880966666 at 195.189.173.27 Action bridge(1000 XML
default)

 

This means somewhere in your dialplan you must have

 

<action application="bridge" data="1000 XML default"/>

You are using the syntax for the transfer app in the bridge function.  Maybe
this is one of the known errata in the book if this is where you got the
example.

 

try transfer instead of bridge





On Fri, Oct 12, 2012 at 10:20 AM, Richard Weremiuk <richwerem at gmail.com>
wrote:

 

Good afternoon

 

I'm new to freeSWITCH and have setup my freeSWITCH successfully, from the
latest pre-compiled version (windows).  I have a successfully configured a
SIP gateway (provider voipfone), but when I dial into the PSTN number, just
to try and route to one of the default build extensions 1000 (I also tried
the IVR) I get this error:

 

2012-10-12 15:41:16.997549 [ERR] switch_core_session.c:408 Could not locate
channel type 1000 XML default

2012-10-12 15:41:16.997549 [NOTICE] switch_ivr_originate.c:2599 Cannot
create outgoing channel of type [1000 XML default] cause:
[CHAN_NOT_IMPLEMENTED]

 

I've taken the code straight out of the book, and just added my SIP username
and password, so I can't understand why it won't route through?  Rather than
past in code here, I have put the output from a clean startup in a word
file, and also put the external sip_profile xml, and also the public
directory xml file into the same document too.

 

http://www.beyondmarathon.com/routes/freeswitch_output.docx

 

voipfone was not a provider listed in the wiki, but given I have been able
to get the gateway registered then I assume it is ok, and there must be a
config issue somewhere? 

 

Any help would be greatly appreciated.

 

Regards

 

Richard Weremiuk


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Anthony Minessale II

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