in your sip.cong [general] contexts
put
disallow=all
allow=ulaw
allow=alaw
and in your sip user, use disallow only ONCE, that is
disallow=all
allow=ulaw
allow=alawhope this helps.
regards,
Umair bari
On 12/15/05, Jason Chan (jasonOfficial) [EMAIL PROTECTED] wrote:
Hi there,I am writing to
And if, for some very strange reason, it doesn't work, use noload at modules.conf ;)
Regards,
2005/12/15, Umair Bari [EMAIL PROTECTED]:
in your sip.cong [general] contexts
put
disallow=all
allow=ulaw
allow=alaw
and in your sip user, use disallow only ONCE, that is
disallow=all
allow=ulaw
Hi there,I am writing to ask
about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to
use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450
FXO Port, but this gateway justsimply doesn't support RFC2833 nor SIP-INFO.
The only method I can use