asterisk users - May 2008 - Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

Alright guys and gals,
I'm a little lost, I'm primarily a SIP/IAX based guy, and have ended up 
with a Zap installation. Everything was fine with our old provider when 
we were using PRI, but the new provider screwed up on provisioning and 
we've been temporarily stuck with a pair of EM Wink T's. Ever since 
then, we've been dropping 1-2% of all calls (in or out) and even more 
strange, when a call gets dropped, a phantom call was being generated on 
the incoming side, but only by Asterisk, the T providers (Qwest) say 
they have no records of those calls.

So, my question to you is, has anyone else dealt with a EM Wink T before 
using Asterisk, if so did you experience problems similar to this, and 
finally, if so how did you deal with it?

Here's an outline of our system specs:

Dual 2.3Ghz Athlon
2GB RAM
Asterisk 1.4.16 (Tried 1.4.19 as well)
Zaptel 1.4.10

51 Zap phones connected via SEPARATE TE407 and channel bank
2 EM_W T1's connected via TE407
25 SIP Phones

All calls are being recorded by the Monitor() application, there is no 
timeout on the dial command, I can find NOTHING in the system config 
that would instruct Asterisk to dump the call.
I have spoken with the Qwest technicians who have pulled their call 
records, and they report that we "disconnected the call"....

Any ideas, thoughts? I've reviewed the verbose (full setting, writing to 
file) and see that the far end channel disconnects, and then the near 
end goes into TIMEOUT. I've watched full debug output as well, from 
file, cannot find ANYTHING...

Thanks for any help,
Sherwood McGowan

Hello,

I have quite a bit of experience with E&M Wink T1s, and I have seen
the problem you describe twice. In both cases it was either the
carrier's equipment or the wiring somewhere between the carrier shelf
and your equipment.

In one case it was water in the line that would seem to cause the
problem after it rained, and the other case was bad carrier equipment
at their shelf, once they moved it to another port on another shelf
the problem disappeared.

Good luck,

MATT---


On 5/15/08, Sherwood McGowan <sherwood.mcgowan at gmail.com>
wrote:
> Alright guys and gals,
>  I'm a little lost, I'm primarily a SIP/IAX based guy, and have
ended up
>  with a Zap installation. Everything was fine with our old provider when
>  we were using PRI, but the new provider screwed up on provisioning and
>  we've been temporarily stuck with a pair of EM Wink T's. Ever
since
>  then, we've been dropping 1-2% of all calls (in or out) and even more
>  strange, when a call gets dropped, a phantom call was being generated on
>  the incoming side, but only by Asterisk, the T providers (Qwest) say
>  they have no records of those calls.
>
>  So, my question to you is, has anyone else dealt with a EM Wink T before
>  using Asterisk, if so did you experience problems similar to this, and
>  finally, if so how did you deal with it?
>
>  Here's an outline of our system specs:
>
>  Dual 2.3Ghz Athlon
>  2GB RAM
>  Asterisk 1.4.16 (Tried 1.4.19 as well)
>  Zaptel 1.4.10
>
>  51 Zap phones connected via SEPARATE TE407 and channel bank
>  2 EM_W T1's connected via TE407
>  25 SIP Phones
>
>  All calls are being recorded by the Monitor() application, there is no
>  timeout on the dial command, I can find NOTHING in the system config
>  that would instruct Asterisk to dump the call.
>  I have spoken with the Qwest technicians who have pulled their call
>  records, and they report that we "disconnected the call"....
>
>  Any ideas, thoughts? I've reviewed the verbose (full setting, writing
to
>  file) and see that the far end channel disconnects, and then the near
>  end goes into TIMEOUT. I've watched full debug output as well, from
>  file, cannot find ANYTHING...
>
>  Thanks for any help,
>  Sherwood McGowan
>
>  _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said

...
> we've been temporarily stuck with a pair of EM Wink T's. Ever since
> then, we've been dropping 1-2% of all calls (in or out) and even more 
> strange, when a call gets dropped, a phantom call was being 
> generated on 
> the incoming side, but only by Asterisk, the T providers (Qwest) say 
> they have no records of those calls.
...

I don't know whether this could be related or not but are you set to
loop timing on your incoming phone company T1 port? I have seen timing
issues create some strange issues. 

By the way, we are using incoming EM wink trunks delivered over a T1 and
are not having any issues. We are using Asterisk 1.4.18 with Zaptel
1.4.10.  

Don Pobanz

--
MailDefender Message Security: Click below to verify authenticity
http://www.exchangedefender.com/verify.asp?id=m4FH3AwE015747&from=dpobanz at
hastingsutilities.com

Hello,

The capacity greatly depends on the rate of calls entering and leaving
those conferences.

I see that you do call center systems so I would guess that the rate
would be fairly rapid.

It is really something you have to test and see. Using VICIDIAL in
performance testing mode I have gotten to over 100 conferences on a
similarly equipped server with a very rapid call turnover rate.

MATT---


On 5/15/08, Wai Wu <wkwu at calltrol.com> wrote:
>
>  Hi all,
>
>  What is maximum number of three party conferences can a quadcore 3GHz
>  system can handle? All the parties a setup with G.711 codec.
>
>  _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

In many call center applications, conferences are usually long and only
a small number them are necessary in any given time.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
Florell
Sent: Thursday, May 15, 2008 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Number of meetme conferences

Hello,

The capacity greatly depends on the rate of calls entering and leaving
those conferences.

I see that you do call center systems so I would guess that the rate
would be fairly rapid.

It is really something you have to test and see. Using VICIDIAL in
performance testing mode I have gotten to over 100 conferences on a
similarly equipped server with a very rapid call turnover rate.

MATT---


On 5/15/08, Wai Wu <wkwu at calltrol.com> wrote:
>
>  Hi all,
>
>  What is maximum number of three party conferences can a quadcore 3GHz
> system can handle? All the parties a setup with G.711 codec.
>
>  _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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What does the timing look like in the zapata.conf file???
E&M is very sensitive to timing slips....

Is it B8ZS, ESF, D4, AMI????

Alex


Snip

Alexander Lopez wrote:
> What does the timing look like in the zapata.conf file???
> E&M is very sensitive to timing slips....
>
> Is it B8ZS, ESF, D4, AMI????
>
> Alex
>
>
> Snip
>
>
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>   
/etc/zaptel.conf
*snip*
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
e&m=1-48
*snip*

Anyway to disable the join conference notification on the party that is
joining, but not the parties already in the conference?