does anyone of you know wether asterisk can handle SIP_INFO on pure sip
calls? Is that something I have to handle in the extensions? Does
asterisk hand incoming SIP_INFO over to an already connected peer?
Thanks and regards,
Christophorus Laube
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
No comments:
Post a Comment