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VoIP / Asterisk / rtcomm
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Moonshine
2007-11-30 , 19:55
Posts: 469 | Thanked: 88 times | Joined on Sep 2007 @ Montana
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18
Lots of variable here, but you might want to try adding:
canreinvite=no
Then Asterisk won't allow the SIP clients to issue additional invites where they will try to talk directly with each other. (Which can create issues NAT/public addresses, rport, etc)
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