Vicky wrote:
> I have around 20-30 softphones behind NAT .. My sip.conf has
nat=yes and
> they all are able to register and make calls with no problem . My
voip
> carrier supports gsm as well as ilbc .. Server takes calls from sip
> phones ,
> does call recording in between and forwards to voip carrier . My
> problem is
> that half of my softphones use ilbc and rest use gsm and my
provider
> supports both gsm as well as ilbc . Now when i put
allow=gsm&ilbc in my
> voip carrier's extension then it uses gsm ( first preference ) to
send
> calls
> but half of my softphones use ilbc so asterisk does codec
transcoding in
> between using lot of cpu .. how ever my carrier does support ilbc
> tooo but
> when i put allow=ilbc&gsm then it uses ilbc again and does codec
> transcoding
> from gsm to ilbc for rest of softphones . How can i make asterisk
to be
> smart in choosing codec .. and use ilbc to voip carrier if
softphone is
> using ilbc or use gsm when softphone is using gsm ( but still should
> do call
> recording in between ) .. I am using freepbx for most of
configuration
> btw... Any suggestions ?
>
On 08/12/06, Pavel Jezek < [EMAIL PROTECTED]> wrote:you can try this
patch,
0004825: [patch][post 1.4] New codec negotiation algorithm
http://bugs.digium.com/view.php?id=4825
I'm think, this is one of the most wanted feature,
but unfortunately will not be in asterisk 1.4 and we must wait for 1.6
to be officially supported feature :'(
PJ
On 7 Dec 2006, at 21:29, Vicky wrote:
I am still on asterisk 1.2 branch svn ( afraid of word beta on
server :( ) . I will try out that patch.
Alternatively try setting
${SIP_CODEC}
before you place the call to your provider.
I'd love to hear if it works.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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