This question has come up before.. IIRC its to do with the fact that a SIP call has a uniqe ID apended to it on each call so it doen not play nicely in Gastman.. I guess that the Gastman code should be modified to strip off the unique ID from the SIP channel reference..Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon.
Services like Conference bridge and Musichonhold seem to work ok (I use [EMAIL PROTECTED] and [EMAIL PROTECTED]) for the Icon extensions.
IAX softphone seems to work ok (I use IAX/[EMAIL PROTECTED]) for the Icon extension
But for SIP phones, I use SIP/311 for an extension. But when the phone is used (either dialing out or being dialed to) a new icon pops up on the screen (SIP/311-ferh).
If you have a working Gastaman, can you share your configuration file , please?
Anyone have any documentation on Gastman/Astman?
Thanks
Lee Goodman
Later..
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