[Asterisk-Users] SIP codecs

[Dave Alan Caruana]

i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.

How do I find out which codecs are installed, and how
can I install further codecs? Any suggestions which
would be the right one?

I think hte problem is from the Go2Call side, not the
SJPhone cos I can dial from SJPHone to SJPhone
routing through asterisk with no problems.

many cheers
Dave