[cisco-voip] Call routing rules for AS5XXX connected to a PBX

[Jared Mauch]

On Mon, May 23, 2005 at 06:26:25PM -0400, Robert Boyle wrote:
> 
> Hello,
> 
> We use AS5XXX boxes as SIP<->PSTN gateways. Until now, everything has been 
> fine and works ok. Now I have thrown a monkey wrench into the works. I am 
> going to configure one port on one of our gateways to interface with our 
> PBX. When a call comes in via PSTN, I want to route it to our SIP server, 
> unless it matches a list of numbers should go directly to the PBX. When a 
> call comes in via SIP, I want it to go out to the PSTN unless it is 
> destined for the PBX, then it should go directly there. Is is possible with 
> ACLs or dialer lists? Does anyone have a working example they wouldn't mind 
> sharing? Thanks!

	I think you want the right dial-peers to be setup.

	eg:


dial-peer voice 1000 voip
 destination-pattern 1...
 session protocol sipv2
 session target ipv4:10.2.3.4
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 214 pots
 destination-pattern 1214.......
 port 1/0:23
 prefix 214
!

	This will do a few things, 1) take extens 1XXX and send them
to a SIP peer/proxy at 10.2.3.4 for routing.  This could just as
easily be a 'port 1/0:23' which would go to your PBX to dump
4 digit extens there.

	It will also take calls destined to 1214NXXYYYY and prefix them
with 214 and send them out port 1/0:23 (PSTN).

	You want the dtmf-relay as it's the most compatible/standard
way.

	The 'vad/codec' settings are optional, but i've found this to work
the best in a non-bw limited environment.

> This is what I think we should use on the "PRI" port to our PBX - any 
> comments or suggestions?
> --------------------
> interface Serial4:23
> no ip address
> isdn switch-type primary-ni
> isdn protocol-emulate network
> isdn incoming-voice modem
^^^^^^^^^^^^^^^^^^^^^^^^^^^

	I suspect you want to change that to be
voice.. (not modem)

> isdn guard-timer 3000
> isdn T310 120000
> isdn negotiate-bchan resend-setup

	You may want to turn on some interface logging as well:

interface Serial1/0:23
 logging event link-status
 logging event nfas-status
 logging event subif-link-status

If you're doing 100% voice, you may find these helpful too:

interface Serial1/0:23
 isdn voice-priority always
 isdn incoming-voice voice
 isdn reject data

Under the global config, you may want to have some
logging/debugging available, here's my recommended settings for a pure
SIP/POTS gateway:

isdn logging
isdn-mib retain-timer 336
!
voice service pots 
!
voice service voip 
 h323
  call service stop
 sip
!
call-history-mib retain-timer 500
call-history-mib max-size 500


	- jared

-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.