Please do not post same thing again and again . It wont help you get better replies , Post you asterisk cli output while call is in progress and when it disconnects prematurely .
On 18/06/07, Don Kelly <[EMAIL PROTECTED]> wrote:
I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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