AsteriskList [EMAIL PROTECTED] wrote: > what does the command NOTRANSFER in IAX.CONF? > where do i find asterisk�s commands? > In the website, VOIP-INFO.ORG I did not find anything regarded to > NOTRANSFER commands unfortunatly. > If transfer is allowed then a call may be "transferred" from one IAX2-driven service to another in order to reduce the number of hops between endpoints. If the endpoints are both IAX2-driven then they could automatically transfer to talk directly to one another. Notransfer prevents this and keeps the Asterisk server "in the loop".
The same is possible with SIP. See "canreinvite" in sip.conf, which can be set to "no" to prevent the actions described above (when using SIP instead of IAX2, of course). IAXtel (when it works) allows transfer to keep itself out of the loop and to allow the endpoints to talk directly. FWD allows canreinvite (SIP mode) and allows transfer (IAX2 mode) to reduce its bandwidth usage to the bare minimum. If none of this is in the Wiki (I didn't check) then feel free to add it. There are other ways to keep Asterisk in the loop and prevent transfer or re-invitation. If the 'T' or 't' flags are passed to Dial() then Asterisk will have to remain in the loop to listen for the '#' tone. I know that this will prevent a re-invitation in SIP. I imagine the same is true for IAX2 transfers. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
