Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-20 Thread Sherwood McGowan
Thank you all for your input. Currently, nothing has improved the 
dropped call rate by more than .2%, leaving me at 1.8% dropped calls 
still..Luckily, our switch back to PRI is due anytime in the next day or 
so..

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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-16 Thread Sherwood McGowan
Eric Wieling wrote:
 Make SURE you are not using callprogress=yes or busydetect=yes (they 
 default to no).  These options are commonly known in the Asterisk world 
 as randomlydisconnectmycalls=yes.

 Sherwood McGowan wrote:
   
 Steve Totaro wrote:
 
 On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
 [EMAIL PROTECTED] wrote:
   
   
 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[EMAIL 
 PROTECTED]


 
 
 How did your dialplan change.  When do the hangups occur?  Is there a 
 pattern?

 I have done a few EM wink setups with no issues other than getting
 the configs and dialplan right.

 Personally, I would ream your provider rep and demand that your
 situation be escalated to the top.  If they don't agree, go over their
 head, I always get an escalation form with contacts for each
 escalation level.  Obviously, when thing are straightened out, make it
 a point be very thankful to everyone and CC them with what a great job
 they did.

 Thanks,
 Steve Totaro

   

   
 Steve,
 Thanks for the input. The dialplan did not change when this symptom 
 started happening. There is no pattern other than we drop 1 or 2 calls 
 at a time, I've reviewed the dialplan executions (verbose level 3 output 
 to log file) and the call is executing as normal but then the far end 
 (trunk side) just disconnects, and we then go to timeout (which it should).

 I've tried to be cool with our reps so far, but it IS getting a little 
 ridiculous, sounds like I may have to go that route.

 Sherwood McGowan
 


   
Thanks for the tip. I had those enabled, but not for the EM_W trunks, 
only for my POTS end. After speaking with Digium I have disabled them 
completely, and I'm waiting to see the results from today's set of calls.

Thanks to everyone so far for the suggestions, it's good to know that 
even after all these years (and an absence from the mailing lists) I 
still find the community just as helpful and kind as before :)

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


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

2008-05-15 Thread Sherwood McGowan
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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Matt Florell
Hello,

I have quite a bit of experience with EM 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 [EMAIL PROTECTED] 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


___
-- 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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Sherwood McGowan
Matt Florell wrote:
 Hello,

 I have quite a bit of experience with EM 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 [EMAIL PROTECTED] 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

 

 ___
 -- 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
   
Thanks for the info, I'll see what I can figure out. I'll triple check 
the wiring here on our end, and try to figure out how I can convince 
Qwest to try putting us on another port on a different shelf

Cheers,
Sherwood

___
-- 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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Don Pobanz
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[EMAIL PROTECTED]



___
-- 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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Steve Totaro
On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
 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[EMAIL PROTECTED]



How did your dialplan change.  When do the hangups occur?  Is there a pattern?

I have done a few EM wink setups with no issues other than getting
the configs and dialplan right.

Personally, I would ream your provider rep and demand that your
situation be escalated to the top.  If they don't agree, go over their
head, I always get an escalation form with contacts for each
escalation level.  Obviously, when thing are straightened out, make it
a point be very thankful to everyone and CC them with what a great job
they did.

Thanks,
Steve Totaro

___
-- 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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Sherwood McGowan
Matt Florell wrote:
 Hello,

 I have quite a bit of experience with EM 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 [EMAIL PROTECTED] 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

 

 ___
 -- 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
   
I'll check that out, as far as I know I'm not, although our Rhino 
channel bank uses loop timing. Another thing I'm worried about is my 
zttest results have been dipping VERY low, sometimes down to 95% (worst 
hit, not avg)but this server should be able to handle this 
EASILYwe didn't have the accuracy problems when on PRI, and nothing 
has changed other than that. Quick note, I've checked and we're not 
missing interrupts, and I've gone CRAZY with checking all our Zap 
settings with Digium and with Qwest, to no avail.

I'll check the wiring here in the office tonight to make sure the guy I 
inherited this from didn't make a mistake somewhere, and act on anything 
else you guys might have.

Thanks so far, you're all helping me at least think of new things to 
check out. I've barely worked with Zap, and mostly only on the local 
side with channel banks.

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


Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?

2008-05-15 Thread Sherwood McGowan
Steve Totaro wrote:
 On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
 [EMAIL PROTECTED] wrote:
   
 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[EMAIL 
 PROTECTED]


 

 How did your dialplan change.  When do the hangups occur?  Is there a pattern?

 I have done a few EM wink setups with no issues other than getting
 the configs and dialplan right.

 Personally, I would ream your provider rep and demand that your
 situation be escalated to the top.  If they don't agree, go over their
 head, I always get an escalation form with contacts for each
 escalation level.  Obviously, when thing are straightened out, make it
 a point be very thankful to everyone and CC them with what a great job
 they did.

 Thanks,
 Steve Totaro

 ___
 -- 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
   
Steve,
Thanks for the input. The dialplan did not change when this symptom 
started happening. There is no pattern other than we drop 1 or 2 calls 
at a time, I've reviewed the dialplan executions (verbose level 3 output 
to log file) and the call is executing as normal but then the far end 
(trunk side) just disconnects, and we then go to timeout (which it should).

I've tried to be cool with our reps so far, but it IS getting a little 
ridiculous, sounds like I may have to go that route.

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