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

Jeremy Hinton lists at cablemonkey.com
Mon Mar 21 11:36:55 EST 2005


Adam Rothschild wrote:
> Hello,
> 
> I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in
> turn is talking to an Asterisk server via SIP for call origination and
> termination.  Seems simple enough, and it works for the most part,
> but:
> 
> 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.

I've had not luck with this either. Looks like asterisk (atleast the one 
i'm using) can't understand the messages from the cisco, as i'm seeing 
this in my * logs right after the call:

Mar 21 09:33:24 WARNING[196620]: chan_sip.c:6116 receive_info: Unable to 
parse INFO message from DDB5CDF5-994D11D9-98DB93DC-DDF2614 at 209.96.249.5. 
Content

> 2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk
>    -> Dial([...],,r) calls placed.  Music on hold ([...],,m) works
>    fine.

Could be that your calls are pstn hairpinning due to a dial peer mixup. 
Saw this on our 2621 here, until i revised my dial peers. do a "debug 
isdn q931" on the cisco (presuming the same command as our 26xx), if you 
see the cisco sending the call back out the pri, thats what happening. I 
  got around this using CoS in the dialplan. If this is the case, let me 
know and i can send you my solution.

- jeremy

-- 
Jeremy Hinton                     A little nonsense
Senior Network Manager               now and then
Continental VisiNet Broadband       is relished by
jgh at visi.net                        the wisest men
757 873 4500


More information about the cisco-voip mailing list