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

[Steven Critchfield]

On Mon, 2003-06-30 at 00:35, Andy Hester wrote:
> Steven,
> 	I thought that "1" would mean that my T100P card would set the timing for
> the line.  Is this incorrect?  If I am reading this wrong then please set me
> straight.
> 
> My carrier has their end set to be the sync source.  If I set the timing to
> "1", won't that conflict?

Not really, a 1 signifies the span is the primary sync source. This
would be important when you have multiple spans to determine which one
to get timing from and then use it for the other spans.  
BTW, I used it this way when I did similar drop and inserts on a zhone
unit to my asterisk machine and I used a 1 for my timing.

> My line is set up esf/b8zs, so does that mean I can ignore all bipolar
> violations, or just that a certain number are to be expected?

Usually the ADIT will let you know when it is incorrect. Legitimate
bipolar violations occur only during certain conditions and the
equipment should be able to determine when that is true.

> 	Also, shouldn't the switch tech from my carrier be knowledgeable about
> these things and trying to help me match up to their settings?

The telco should be knowledgeable about this, but since your link to
asterisk terminates in the ADIT their switch shouldn't see the problems.
The ADIT should insulate them from any number of problems, and I doubt
XO is looking at the information on the ADIT.

BTW, I know XO uses ADIT 600s at customer sites.


> 	I appreciate you indulging me this evening.  I'm in somewhat of a bad spot
> with my customer for a variety of reasons, most beyond my control and I am
> trying to get their problems resolved asap.  Thanks again.
> 
> Sincerely,
> Andy Hester
> Consero
> 
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Steven
> > Critchfield
> > Sent: Monday, June 30, 2003 12:04 AM
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] Help! Problems talking to upstream switch
> >
> >
> > 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>
> > >
> > > _______________________________________________
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > --
> > Steven Critchfield <critch at basesys.com>
> >
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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-- 
Steven Critchfield  <critch at basesys.com>