[Asterisk-Users] sipura spa2 + asterisk bug ?

[Tofik Suleymanov]

Steve Kennedy wrote:
> On Tue, Mar 28, 2006 at 07:40:06PM +0000, Tofik Suleymanov wrote:
> 
> 
>>Each of the two lines have their own entry in sip.conf and i can see 
>>each line registered in 'sip show peers'.
>>I can dial each line from outside successfully but when one line is busy 
>>i can't reach the second line (it immediately sends me to the 
>>voicemail).I've also tried to change the timeouts in dial command but 
>>seems that it doesn't matter.
>>Any other advice ?
> 
> 
> You haven't got codec negotiation set-up properly so it's still running
> out of g.729 and then it will act as busy
> 
> I have
> 
> dtmfmode=rfc2833
> disallow=all
> allow=g729
> allow=gsm
> allow=alaw
> allow=ulaw
> allow=g723.1
> 
> 
> So should try g.729 first, then gsm (which unfortunately SPA don't
> support), etc etc.
> 
> 
> Steve
> 
Thank you very much !
after playing a bit with codecs i've managed my sipura lines to work 
properly.

Again, thank you very much for quick and effective help !

Tofik Suleymanov