[asterisk-dev] SIP trunking

[Anton]

On 3 March 2007 14:08, Tilghman Lesher wrote:
> > before anything start happening, it should be
> > expressed, questions asked. The question is about
> > "you're listening, but not hearing".
>
> No, I heard it very plainly.  You want your provider
> supported, but you are unwilling or unable to code it
> yourself.  It's fine that you would like this coded.  I
> would suggest that you post a bounty on the Wiki.

This I mean. It may be interpreted anyhow, and actually I 
don't care that is not supported, since anyway will get it 
my way much sooner that when it could be coded. One of the 
way is trying placing SipX in between. Other is play with 
OpenSS7, which supports Sip-T - third is to get their 
support request this feature from SoftX - the easiest, what 
will be done first. 
But as the one who believes in OpenSource - I would be happy 
to see that feature in Asterisk and if needed provide a 
testbed for someone wish to test interoperability with 
carrier softswitch. Yes I can code it myself either... More 
correctly to say "I'm able" since will not have enough time 
to do it anyway, so, cannot, otherwise I would. Don't want 
to argue on this, none cares or no interest - let it be 
so...