Sorry, Good news is we are doing a complete refresh of the book to correct every known error from the last 3 years and update it to match 1.2! On Fri, Oct 12, 2012 at 12:44 PM, Richard Weremiuk <richwerem@gmail.com>wrote:
Anthony****
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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.****
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That's really helped. Pulled my hair out all day with that.****
Richard****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@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****
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Dialplan: sofia/external/07880966666@195.189.173.27 Action bridge(1000 XML default)****
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This means somewhere in your dialplan you must have****
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<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.****
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try transfer instead of bridge****
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On Fri, Oct 12, 2012 at 10:20 AM, Richard Weremiuk <richwerem@gmail.com> wrote:****
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Good afternoon****
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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:****
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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]****
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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.****
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http://www.beyondmarathon.com/routes/freeswitch_output.docx****
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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? ****
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Any help would be greatly appreciated.****
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Regards****
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Richard Weremiuk****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900