> I discovered some serious problem with several Sipuras 3000 but I don't know > if the problem is with them or Asterisk. Basically, if I call a Sipura > PSTN line, when there is a call already in progress, generally I get a 503 > Sevice Unavailable, but if I try hard enough, I am able to get through > and connect to dialed number. The other call gets disconnected but the > originator of the other call is now on my call. Is this a bug of Asterisk's > SIP implementation? or is it a Sipura configuration problem? > > I looked at other alternatives but haven't had any luck. Hint didn't work and > CheckGroup does exactly the same thing. Sometimes I get Service > Unavailable but other times i can dial even though there is a call in > progress.
Its almost impossible to respond to the above since the spa3k can be configured in lots of different ways, and can interact with * in lots of different ways. Since Sipura does not provide a nice way to post their configs, it makes it even more difficult. Personally, I've been using a spa3k with * for about nine months, and really haven't had any problems with it. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
