I am having trouble setting the /etc/asterisk/sip.conf file. This is my file:
1) Add in the [general] section:
disallow=all allow=ulaw allow=alaw allow=any other codec that you want to (or can) support.
While some have found that this must be specified for each and every phone, I have found that it works fine specified just once in the general section.
2) Include dtfmmode=info or inband and match to phone's setting[243075] type = friend context = default secret = gol host = dynamic callerid = fono75 <243075>
3) I may have been too tired at the time, but once I tried using long extensions (more than 5 digits) and could not make them work either - same error you are getting. I would limit your extensions to 4 digits and see if it helps.
4) You may also need to add
canreinvite=no
to each phone definition.
and our SIP phones configuration are the following:
SIP Server: 192.168.0.102
Outbound Proxy: <Empty>
5) I would set this to be the same as the server if you want to make outbound calls.
Hope this helps
Stephen R. Besch _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
