I have two ATA186s talking to an asterisk server. When I call in on an outside line, both ring, and I can pick up either and talk.
But if I try to call from one of them to the other, the remote end rings just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a "Call/Leg Transaction Doesn't Exist" error and hangs up the line.
What am I doing wrong here?
Thanks.
B.
Here are the respective sip.conf entries:
[ata3] type=friend username=ata3 secret=asecret host=129.91.0.164 context=home
[sjcata] type=friend username=sjcata secret=anothersecret host=129.91.0.161 context=home
In my extensions.conf:
exten => 83,1,Dial,SIP/sjcata|25 exten => 83,2,Voicemail,u100 exten => 83,3,Hangup
exten => 84,1,Dial,SIP/ata3|25 exten => 84,2,Voicemail,u100 exten => 84,3,Hangup
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