Thanks John, we managed to get it going after creating another trunk and
rebooting the server.
It seems that changes made to the trunk info. do not take affect until a reboot.
Also any changes to SIP.CONF get overwritten by Freepbx.
The SIP information is traditionally #INCLUDED in other files like sip_nat.conf
or sip_general_custom.conf
(preferred) which do not get rewritten by Freepbx.
Thanks
H
=================================
Henry L.Coleman [www.VoIP-PBX.ca]
Tel: 647-723-5160 Ext.203
=================================
{ John Lange}
> On Wed, 2009-02-04 at 04:35 -0500, Henry L.Coleman wrote:
>> I'm having all sorts of grief trying to get outgoing calls routed to this
>> SIP provider
>> Does anyone have a working trunk configuration that I can use with Freepbx
>> (Trixbox etc.)?
>> After three days we have got incoming calls working but outgoing proves to
>> be elusive.
>> Hint:
>> Outgoing:
>> Peer Details:
>>
>> context=from-pstn
>> canreinvite=no
>> fromdomain=(domain name)
>> host=(ipaddress)
>> insecure=invite,port
>> type=friend
>> disallow=all
>> allow=ulaw
>
> We don't use then for any inbound but here is our outbound sip.conf. I
> Believe they also have examples for Asterisk on their web site no? If
> not you can email their support.
>
> [thinktel]
> type=friend
> host=vp.thinktel.ca
> nat=no
> context=from_thinktel
> dtmfmode=rfc2833
> canreinvite=no
> qualify=no
> disallow=all
> allow=ulaw
> ;allow=g729
>
> --
> John Lange
> www.johnlange.ca
>
>
>
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