register on gateway.freshtel.net, not cts-au
-for firefly numbers, call gateway.freshtel.net
-for PSTN termination, call cts-au.freshtel.net
don't think you need the @freshtel at the end of your dial
good luck,
Adam
Shaun Dwyer wrote:> Hi,
>
> I'm having some problems getting calls to go out via freshtel.
> There dosn't seem to be any specific information on how to get it
> working anywhere.
>
> The only information I've found is here:
> http://www.voip-info.org/wiki-Freshtel
> and that dosn't give you any idea of how to actually get it working.
>
> I've tried adapting information from other IAX2 provider examples but
> have yet to find a working solution.
>
> in my iax.conf I have:
>
> register => freshtel_number:password@cts-au.freshtel.net
>
> [freshtel]
> type=friend
> host=cts-au.freshtel.net
> secret=password
> context=from-freshtel
> qualify=yes
>
> In my extensions.conf, I have:
>
> exten => _99.,1,StripMSD,2
> exten =>
> _99.,2,(Dial(IAX2/freshtel_number:password@freshtel/${EXTEN}@freshtel)
>
>
>
> The general idea is to dial 99 followed by the number to dial thru
> freshtel.
>
> In my SIP client phone (X-Lite) I get 'call failed: 403 Forbidden'
>
> on the asterisk server console I get:
> -- Executing StripMSD("SIP/101-b0d9", "2") in new
stack
> Sep 7 14:42:55 WARNING[1107577776]: pbx.c:1872 ast_pbx_run: Channel
> 'SIP/101-b0d9' sent into invalid extension '88669100' in
context
> 'intern', but no invalid handler
>
>
>
> I have multiple SIP phones, all can dial eachother as well as the echo
> test extension I've setup. They can also
> interact with voicemail with no problems.
>
> I also have a X101P card setup connected to a PSTN line and I can make
> calls though that OK as well.
> There is an echo problem with the X101P, but thats another story...
>
> Any one have any suggestions with regards to my freshtel problem?
>
> I've yet to try a SIP connection to them. I'll be trying that later
> thisarvo.
>
> Cheers,
> -Shaun
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