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

[Steven Critchfield]

Use 1 for timing. You actually do have a timing source to sync to.

As for slips and bipolar violations...
T1s are just high speed serial lines. A sleep is when you loose sync
with the far side and when you see a 1 come across the line, you may not
know which bit it was for. This would be a slip. Bipolar violations are
a part of the signaling, but can also be errors. A T1 alternates the
polarity of the 1 pulse to allow the line to run farther on lower
voltage. Just doing alternating polarity is AMI or Alternate Mark
Inversion. A bipolar violation is when a bit is received as the same
polarity as the last bit received. On an AMI line a bipolar violation is
an error. On a B8ZS, bipolar violations are intentionally inserted into
the line to keep the line from transmitting too many 0's in a row and
contributing to a slip. When set for B8ZS the "error" is somewhat
expected and ignored.  

On Sun, 2003-06-29 at 23:19, Andy Hester wrote:
> 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>
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield <critch at basesys.com>