[cisco-voip] CISCO(29xx series) Router /Norte 11c Meredianl E1 Connectivity - call drop after 2 min 45 sec

[Humayun Sami/Engineering/Karachi]

I can see the cause 102 error on both sides. I assume when the call is sent to the PBX, it does not get the acknowledgement signals but in that case when I remove .T from the Dial-peer the call does drop but I get no dial tone.

I just don't get the point why only PSTN calls disconnect after 2min 45 sec.

Attached are the debugs, I hope it helps.


Regards,
________________________________
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Friday, April 15, 2011 8:58 PM
To: Humayun Sami/Engineering/Karachi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CISCO(29xx series) Router /Norte 11c Meredianl E1 Connectivity - call drop after 2 min 45 sec

Which side initiates the disconnect, and can you paste the sanitized q931 debug from the router with the E1?

-Ryan

On Apr 15, 2011, at 7:32 AM, Humayun Sami/Engineering/Karachi wrote:


I have following setup configured: Nortel PBX---E1-->CISCO Router----->CISCO Router (FXO/FXX Port) --- Having problem of dropping call exactly after 2 minutes and 45 second. Analog set located at remote site use 9 to access PSTN trunk installed at Nortel PBX and then it dials out. These calls are dropping after 2 min. 45 seconds...

The CISCO logs and the Nortel 11C capture shows following error
Cause No. 102 - recovery on timer expiry. This cause indicates that a procedure has been initiated by the expiration of a timer in association with error handling procedures.
What it means: This is seen in situations where ACO (Alternate Call Offering) is being used. With this type of call pre-emption, the Telco switch operates a timer. For example, when an analog call is placed to a Netopia router that has two B Data Channels in place, the router relinquishes the second channel, but if it doesn't happen in the time allotted by the switch programming, the call will not ring through and will be discarded by the switch.
<http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php>

   ISDN Serial0/3/1:15 Timers (dsl 0) Switchtype = primary-qsig
       ISDN Layer 2 values
        K     =   7 outstanding I-frames
        N200  =   3 max number of retransmits
        T200  =   1.000 seconds
        T203  =  10.000 seconds
       ISDN Layer 3 values
        T302  =  15.000 seconds
        T301  = 300.000 seconds
        T303  =   6.000 seconds
        T304  = 120.000 seconds
        T305  =  30.000 seconds
        T306  =  30.000 seconds
        T307  = 180.000 seconds
        T308  =   6.000 seconds
        T309    Disabled
        T310  = 120.000 seconds
        T313  =   6.000 seconds
        T314  =   6.000 seconds
        T316  = 120.000 seconds
        T318  =   4.000 seconds
        T319  =   4.000 seconds
        T322  =   4.000 seconds
        T3OOS =   5.000 seconds



Regarding the Dial-peers when I use the below dial peers:

No dial tone but call does not drop

dial-peer voice 82 pots
 description ** PTCL Dialing **
 destination-pattern 82
incoming called-number .
 direct-inward-dial
 port 0/3/1:15
 forward-digits all


Dial Tone but call Drops in 2 min 45 sec

dial-peer voice 82 pots
 description ** PTCL Dialing **
 destination-pattern 82T
incoming called-number .
 direct-inward-dial
 port 0/3/1:15
 forward-digits all

Kindly help...


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