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voicetraffic
2003 Apr 24
7
Outgoing SIP Call to unregistered Users
...t each client, so client A sends to
port 32000 but client B listens on port 32002. One solution for this problem
ist to use the canreinvite=no statement in sip.conf, but in this case every
rtp-packet is going through asterisk. I think, only the SIP/SDP packets
should go through asterisk and the voicetraffic direct from client A to
client B. May be, I'm wrong about that, please correct me in that case.
Another problem is calling SIP users that are not registered to asterisk.
Giving kphone the address sip:name@anyhost causes asterisk to search
for the extension name, but there is no such exten...