[asterisk-users] SIP Peering--call terminated prematurely

[Don Kelly]

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

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