search for: asterisk_b

...ll clears up. Ending the A to B call first does not improve the A to C call. The dialplans are setup so each server passes all non-local extensions to it's neighbor. Hence, for A, the relevant part of the dialplan is exten => _2XXX,1,Verbose(1|Extension 2xxx) exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _2XXX,n,Hangup() exten => _3XXX,1,Verbose(1|Extension 3xxx) exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _3xxx,n,Hangup() For B: exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN}) exten => _1XXX,n,Hangup() exten => _3xx...

...oviders and its main purpose as a call termination forwarder. Asterisk B: it?s connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic going between them, Asterisk_Main will send the provider IP address where both Asterisk_B & provider will communicate, at the end of the call, Asterisk_Main will log CDR traffic. Now I can make IAX2, but the problem is, the traffic must go like this: Provider <->Asterisk_Main<-> Asterisk_B This will cause a problem with wasted bandwidth and more latency on the...

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed