i have s asterisk 1.2 box installed, got few confusions need to be resolved, thought it probably sound stoopid.
i have few notebooks which on the move setup with softphone, do 2 notebooks locate at 2 different remote locations consume asterisk server site bandwidth when they talk to each other?(do the 2 system talk to each other directly after established the connection?)
regarding extension, if users call/transfer mistakenly a extension which doesn’t exist, what is the best practice to remedy for this scenario, i notice if user need to do a call transfer, he pressed a wrong number, he was not able to correct it or pick up back the original line using his hardphone.
welcome!
- that depends. SIP has the capability to make them talk directly to each other using a function called reinvite. If sip.conf has set canreinvite=no in the [general] section, or if the phone’s individual channel has canreinvite=no set, then the reinvite will not occur and asterisk will continue to process the media stream (thus using 2x the call bandwidth at the asterisk server, once in, once out). Keep in mind that reinvites can cause problems with NAT traversal, especially if everything else isn’t set up exactly right. Make sure that the clients behind NAT use STUN correctly, have ports forwarded if applicable, and if the Asterisk server is behind NAT define localnet= and externip=. Set nat=yes for all NAT’d clients.
for transfer correction, you could at least make sure there is an i extension which will absorb the failed transfer and send them back to your main IVR or receptionist, it’s better than nothing, although there is a better way of doing it I can’t think of it at the moment.