Hi Folks,

I'm pretty rusty on my H.323 gateways and dial peers etc, running into a
problem with a configuration in my lab that i'm hoping to use this Fall for
a project - was hoping someone might have an idea.

Goal here is to have a router with CME setup for "SRST Mode" and a number of
POTS lines in from pstn. Phones normally registered to callmanager (version
4.2 in my lab). During normal operation, if any of those lines ring, it
should just connect to call manager and ring a shared line appearance on X
number of phones at the location (probably about 8). During SRST operation,
essentially I want to do the same thing - using overlays etc.

I have the CME up and seems to work OK. The problem I have is during normal
operation. Any 1 line can ring in and it gets sent to call manager and the
correct line rings on all phones. Any other line that rings in after that
appears to get sent to call manager, the line rings once, and then the call
drops. It seems to me that perhaps something in the feature negotiation
phase is failing, since the call rings out once before failing. I dont have
any CAC setup, and its all using G.711ulaw. I don't know enough about H323
to know where to look next though - nothing in the debugs I looked at jumped
out at me as an error.

I did watch the debug for dial-peer matching and it looks to be matching my
dial-peer out to CCM before it matches the ephone-dn dial peers

router is basically setup like this:

voice-port 0/0/0
connection plar 37111
voice-port 0/0/1
connection plar 37111
voice-port 0/0/2
connection plar 37111
!
dial-peer voice 100 voip
destination-pattern 37111$
session target ipv4:10.182.140.10  ! Call Manager 4.2
no vad
codec g711ulaw
!

I found that if I dont put the end of string marker in my destination
pattern, it likes to match the ephone-dn's before it matches the callmanager
dial-peer.

Thanks!
--
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations

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try to find out which end is dropping (releasing) the call (Gateway or
CallManager) and for what reason (h225 q931 cause code)

use "debug h225 q931" on the gateway to get this info...

Regards
Patrick

On 8/16/07, Ed Leatherman <ealeatherman@gmail.com> wrote:
> Hi Folks,
>
> I'm pretty rusty on my H.323 gateways and dial peers etc, running into a
> problem with a configuration in my lab that i'm hoping to use this Fall fo
r
> a project - was hoping someone might have an idea.
>
> Goal here is to have a router with CME setup for "SRST Mode" and a number 
of
> POTS lines in from pstn. Phones normally registered to callmanager (versio
n
> 4.2 in my lab). During normal operation, if any of those lines ring, it
> should just connect to call manager and ring a shared line appearance on X
> number of phones at the location (probably about 8). During SRST operation
,
> essentially I want to do the same thing - using overlays etc.
>
> I have the CME up and seems to work OK. The problem I have is during norma
l
> operation. Any 1 line can ring in and it gets sent to call manager and the
> correct line rings on all phones. Any other line that rings in after that
> appears to get sent to call manager, the line rings once, and then the cal
l
> drops. It seems to me that perhaps something in the feature negotiation
> phase is failing, since the call rings out once before failing. I dont hav
e
> any CAC setup, and its all using G.711ulaw. I don't know enough about H323
> to know where to look next though - nothing in the debugs I looked at jump
ed
> out at me as an error.
>
> I did watch the debug for dial-peer matching and it looks to be matching m
y
> dial-peer out to CCM before it matches the ephone-dn dial peers
>
> router is basically setup like this:
>
> voice-port 0/0/0
>  connection plar 37111
> voice-port 0/0/1
>  connection plar 37111
> voice-port 0/0/2
>  connection plar 37111
> !
> dial-peer voice 100 voip
>  destination-pattern 37111$
>  session target ipv4:10.182.140.10  ! Call Manager 4.2
>  no vad
>  codec g711ulaw
> !
>
> I found that if I dont put the end of string marker in my destination
> pattern, it likes to match the ephone-dn's before it matches the callmanag
er
> dial-peer.
>
> Thanks!
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>  ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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