[Asterisk-Users] Help! Problems talking to upstream switch

Andy Hester cgadmin at conserogroup.com
Sun Jun 29 21:19:04 MST 2003


Thanks for the info... I've answered your questions below.  I am not
experienced with telecom at this level (yet), but this sounds like really
good info to quiz XO's switch tech over.

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Steven
Critchfield
Sent: Sunday, June 29, 2003 9:40 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Help! Problems talking to upstream switch


Whats the sound quality like on the calls especially when multiple calls
are going?

No problems with sound quality save the slight echo on calls over the TDM
circuit.


On my home system, I had a problem that DTMF was sometimes not correctly
recognized and would either dial incorrectly or not at all. It was
evedent when dial tone was played that it crackled. Also when multiple
calls where running each would start to sound like crap. This was later
tracked down to a timing problem.

Since you mention using a CAC with 2 ports on it and a router, I'm going
to assume you have a ADIT 600. Make sure the Adit is set to take timing
from the telco, and then make sure you are set to take your timing from
it. Check to make sure each of the T1 ports a:1 and a:2 are set to take
timing from a:1 if it is your telco port. This will keep you from
slipping and causing potential problems.

I haven't looked around inside the CAC yet since it is XO's, not sure if it
is an
ADIT 600 but it sounds like the same unit.  I am set to "0" for timing and
"0" for
LBO in Zaptel.conf  I assume this is correct for * and that I need to verify
the a:1/a:2 timing settings in the CAC unit?

Come to think of it, there is a way to test this without bringing the
T1s down. The Adit 600 has a show performance command, I may be wrong,
but I'm sure it was performance, anyways it allows you to see slips and
bipolar violations and a couple other stats. This was beneficial for me
as the T100P didn't report problems but the Adit did.

Can you give me a brief idea of what slips and bipolar violations are?

Home this helps.

I am glad to find that someone knows more than my little knowledge of the
subject!
To here the techs talk you'd think that they'd never run into anything like
this before.


On Sun, 2003-06-29 at 20:59, Andy Hester wrote:
> Hi,
> 	Please let me know if you have any ideas - I am taking wild guesses
now....
> Here is the situation:
>
> 	I put in Asterisk for a local customer.  I have Fractional T-1 with 12
> Voice & 12 Data.  I have a T100P hooked up to a TDM Card (they call it a
> chanel bank although it only has 2 outputs) in a CAC unit.  The unit also
> has a router card that runs the data side.  My extensions are all SIP
phones
> save a few fax machines.  The customer has 7 digit unverified account
codes
> on the trunks for billing purposes.
>
> The Problem:
>
> 	As I watch the console, I see calls coming in for exten "73" or "708" or
> "08" or "730" although most come in correctly (ie "7308").  My carrier has
> verified numerous times that they are sending 4 digits.  I have 40 DID
> numbers that need to be routed and they are all in the 73xx range.  I need
> to know anything that would cause my box randomly not to hear all 4 digits
> on occasion.  Also, I have had trouble with people who dial out getting a
> congestion signal mainly on Long Distance numbers.  The person would dial
> the number 4 or 5 times and get congestion then it might go through.  Both
> of these conditions seem to be happening only about 10-20% of the time.
>
> What I have done:
>
> Moved a T100P card to its own IRQ to prevent problems with interrupts -
Did
> not solve either issue.
>
> On the second try, got the carrier to change the way their switch chooses
> channels for incoming calls to prevent "glare" - This MAY have fixed the
> outgoing long distance issue as it seems to have gone away( although it
> doesn't seem logical to me that this would affect only LD) but did not fix
> incoming calls.
>
> Has anyone else had problems getting all of the digits that the Telco
sends?
>
> Thanks,
> Andy Hester
> Consero
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Steven Critchfield <critch at basesys.com>

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