Handling SIP refers when using a SIP Proxy
Dovid Bender <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CAM3TTh0WcXMerabcAPcX=VbEXVT9AUW4vA8_F0x-3gOp6nximQ@mail.gmail.com> |
Hi,,
When using a SIP proxy to load balance calls how do you make it that a call
on an attended transfer reaches the same Asterisk box every time? I was
told that in later versions of Asterisk there is some "magic" to make it
work correctly when load balancing.
TIA.
Dovid
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