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

[Jaswinder Singh]

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 <dk at donkelly.biz> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070618/2f37a618/attachment.htm