[cisco-voip] Re: Asterisk and Cisco AS53xx/54xx Access Server Platform

[Adam Rothschild]

On 2005-03-19-17:50:52, Adam Rothschild <asr+cisco-voip at latency.net> wrote:
[...]
> 1) Caller ID name data comes in on the PRI, but doesn't appear to get
>    handed off to the Asterisk server via SIP, at least not in any
>    format that Asterisk understands.  Caller ID _number_ works fine.

So, for everybody following along at home, it appears as though one
must configure "isdn supp-service calling name" under the sX/Y:23
interface config in order for the Access Server to recognize caller ID
name in the ISDN facility messages.  This feature is available in
12.3(14)T and 12.3(11)T (and likely earlier builds), but not in
12.3(13) mainline.

Problem is, when running T train code, the Cisco has developed a habit
of continuing to ring a SIP channel, even when a PSTN caller has
disconnected.  This appears to be a result of something having changed
in how the box sends out SIP BYE/CANCEL messages, though I'm a bit
confused as to the particulars.  

Clues appreciated!

Thanks,
-a