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2006 Mar 29
1
Inter-Asterisk Using SIP
I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB) On the RECEIVING Server in SIP.CONF: [OB] type=user