Re: [asterisk-users] What does pct mean?

[Floimair Florian]

No you're reading it wrong.

There are 188K received with no loss, and 16441K transmitted.

Still 8809 does not sound like a percentage to me  so there is something wrong 
with either the label or the value.
From what's in the code, you can see it's clearly a lost Packet count not a 
percentage.
So I guess Pct in this case is short for "Packet".

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Maier
Gesendet: Montag, 12. Februar 2018 17:46
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] What does pct mean?

Hi Carsten,

On 02/11/2018 at 07:46 PM Carsten Bock wrote:
> Hi,
> 
> Lost percent (%)

Are you sure? I'm seeing here:

...Receive. .Transmit..
CountLost Pct  Jitter   CountLost PctJitter  RTT
188K  00   0.000188K   16641K 8809   0.000   0.026

=> This doesn't sound reliable to me: there are 188K packets and 16641K of them 
are lost?! The Pct value is fluctuating between about 6009 and 9009.

Thanks,
Michael


> 
> 
> Am 11.02.2018 19:27 schrieb "Michael Maier" :
> 
>> Hello,
>>
>> could somebody please tell me the meaning of "Pct" as seen in asterisk cli:
>>
>> ...Receive. .Transmit..
>> CountLost Pct  Jitter   CountLost Pct  Jitter RTT
>>
>>
>> Thanks,
>> Michael

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,uxGzivqW6oZ241hxc5A3oz1rZf6JLog-Gi4ziwN-95NzbE_HEndRkD8LLLem_gvmmxd5k_T95J2jepit1IpIkZ2AxkG4RSoADT-AMulX4hxaaQ,,=1
 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,wIdvPI63-ImQqWZFblrmlGJbQ_cbmu31TlSPHYv9kAkHrNRLdIAjL2IUr9sVYxm6piHc0Pf2Zna7zuNOJ2hb4CVzp2WYVDsgZJqAyHt2YkVuGQ,,=1
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] What does pct mean?

[Michael Maier]

Hi Carsten,

On 02/11/2018 at 07:46 PM Carsten Bock wrote:
> Hi,
> 
> Lost percent (%)

Are you sure? I'm seeing here:

...Receive. .Transmit..
CountLost Pct  Jitter   CountLost PctJitter  RTT
188K  00   0.000188K   16641K 8809   0.000   0.026

=> This doesn't sound reliable to me: there are 188K packets and 16641K
of them are lost?! The Pct value is fluctuating between about 6009 and 9009.

Thanks,
Michael


> 
> 
> Am 11.02.2018 19:27 schrieb "Michael Maier" :
> 
>> Hello,
>>
>> could somebody please tell me the meaning of "Pct" as seen in asterisk cli:
>>
>> ...Receive. .Transmit..
>> CountLost Pct  Jitter   CountLost Pct  Jitter RTT
>>
>>
>> Thanks,
>> Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

[Khalil Khamlichi]

maybe extension 3082 has some sort of network issue or maybe hardphone
issue. test on another phone / network plug.

On Feb 9, 2018 2:48 PM, "Stefan Viljoen"  wrote:

> Hi Guys
>
>
>
> I have an issue where a call is picked up from a queue. The caller asks
> the person who answered to attended transfer to extension 3082 (for
> argument’s sake.)
>
>
>
> 3082 picks up the attended transfer and speaks with the outside caller
> picked up initially from the queue.
>
>
>
> A few seconds after 3082 has started speaking to the outside caller
>
>
>
> - 3082’s call goes dead in their handset.
>
>
>
> - The outside caller goes back into the queue, hears queue MOH and gets
> answered by another person in the office as if they are dialing in all over
> again.
>
>
>
> - 3082’s phone starts ringing again after they hang up in puzzlement and
> if they then pick up they speak to another person who is trying to make an
> OUTGOING call in their call center.
>
>
>
> This is for a medium sized call center which (along with 17 other centers
> in the same country) run the same dialplan on Asterisk 1.8.32.3 - only
> happens at this location.
>
>
>
> Literally 100 000+ calls are handled across these 18 centers every day,
> only about 10 or 20 at this one center (with carbon-copy dialplan and SIP
> phone hardware types - Yealink T-21Ps - as at every other branch) keeps
> disconnecting people picked up and transferred from the incoming queue from
> the person transferred to, and then connects them and the transferree to
> other phones - the caller as if he is phoning in AGAIN into the incoming
> queue, the transferred-to person to someone else who is trying to dial out
> once the transferred-to person has hung up after losing the incoming caller.
>
>
>
> Anybody ever encountered something similar? The same dialplan on the same
> Ast version runs fine in 17 other locations, some with ten or twenty times
> more traffic and none of these issues.
>
>
>
> No errors or strangeness apparent in the CLI, verbose log, DTMF log...
>
>
>
> Thanks!
>
>
>
> [image: Description: signature]
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Random, uncommanded blind transfers

[John Kiniston]

I'd suggest doing packet captures on the T21P's themselves at the affected
branches and see if you can catch it happening.

The Yealinks themselves will regenerate DTMF if they get signaled for it.

On Thu, Feb 8, 2018 at 2:19 AM, Stefan Viljoen 
wrote:

> Hi Guys
>
>
>
> I’ve got a situation where an incoming call originated from a trunk
> provider will ring in a call center and be answered by an agent.
>
>
>
> Usually within 15 seconds of the incoming call being answered, it will
> randomly blind-transfer to another extension in the same call center.
>
>
>
> It is as if Asterisk is mis-reading some noise in-band as DTMF and doing
> the transfer, or the caller is emitting DTMF to transfer.
>
>
>
> An examination of CEL records for the relevant call shows the transfer
> taking place, but no party (the caller or callee) tried to transfer. The
> relevant queue application already has only small “t” in the parameter list
> (e. g. only let the called user - e. g. my agent inside - transfer the
> incoming call), thereby preventing the caller maybe emitting DTMF down the
> line and transferring at their behest.
>
>
>
> Yet the random, uncommanded blind transfers still take place, always
> withing about 15 seconds after call initiation in answering an incoming SIP
> call.
>
>
>
> We’re using YeaLink SIP phones - T21P - got 17 branches, this only happens
> at 3 of them at random times. Same hardware and Asterisk versions
> (1.8.32.3) at all of them, same dialplans.
>
>
>
> Operating over about several years, about 10 000 000 incoming calls
> handled so far, and this only started happening now... dialplans are
> untouched for months.
>
>
>
> Any ideas, pointers, anybody encountered this before?
>
>
>
> Thanks
>
> [image: Description: signature]
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

[Stefan Viljoen]

Hi Khalil

 

Thanks for the reply. Yup, I’ve switched out the phone with another 
new-from-box Yealink, also moved that network endpoint to another RJ45 port on 
the switch...

 

Regards,

From: Khalil Khamlichi [mailto:khamlichi.kha...@gmail.com] 
Sent: Sunday, 11 February 2018 16:48
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial 
Discussion 
Subject: Re: [asterisk-users] Call picked up from queue and transferred gets 
disconnected - about 0.01% of calls

 

maybe extension 3082 has some sort of network issue or maybe hardphone issue. 
test on another phone / network plug.

 

On Feb 9, 2018 2:48 PM, "Stefan Viljoen"  > wrote:

Hi Guys

 

I have an issue where a call is picked up from a queue. The caller asks the 
person who answered to attended transfer to extension 3082 (for argument’s 
sake.)

 

3082 picks up the attended transfer and speaks with the outside caller picked 
up initially from the queue.

 

A few seconds after 3082 has started speaking to the outside caller

 

- 3082’s call goes dead in their handset.

 

- The outside caller goes back into the queue, hears queue MOH and gets 
answered by another person in the office as if they are dialing in all over 
again.

 

- 3082’s phone starts ringing again after they hang up in puzzlement and if 
they then pick up they speak to another person who is trying to make an 
OUTGOING call in their call center.

 

This is for a medium sized call center which (along with 17 other centers in 
the same country) run the same dialplan on Asterisk 1.8.32.3 - only happens at 
this location.

 

Literally 100 000+ calls are handled across these 18 centers every day, only 
about 10 or 20 at this one center (with carbon-copy dialplan and SIP phone 
hardware types - Yealink T-21Ps - as at every other branch) keeps disconnecting 
people picked up and transferred from the incoming queue from the person 
transferred to, and then connects them and the transferree to other phones - 
the caller as if he is phoning in AGAIN into the incoming queue, the 
transferred-to person to someone else who is trying to dial out once the 
transferred-to person has hung up after losing the incoming caller.

 

Anybody ever encountered something similar? The same dialplan on the same Ast 
version runs fine in 17 other locations, some with ten or twenty times more 
traffic and none of these issues.

 

No errors or strangeness apparent in the CLI, verbose log, DTMF log...

 

Thanks!

 



 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to make International calls from asterisk PBX

[John Novack]

As others have said MUCH more information is needed.
Assume you are using some VOIP provider for international calls
From where to where - gmail gives no clue as to where in the world you are.
Does this provider allow blocking of out of country calls - Do they even 
provide it?
WHICH version of 13?
Care to share a portion of your dialplan?
With your CLI verbosity set high, what are the error messages?
If much of the above is above your paygrade, then perhaps you need to post to 
the biz list and purchase some help?
You will not find any mind readers there either though!


John Novack


Uzma Anjum wrote:

Hello...

I'm running asterisk-13 and international calls are not working in it.How can I 
make it work.Can anyone please tell me.




--

Dog is my Co-pilot

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [OT] Gigaset N510IP provisionning

[Olivier]

Hello,

Has someone met success in Gigaset N510IP DECT base station provisionning ?
If positive, could you describe a bit which files you had to create on
(HTTP) provsionning server ?


Best regards
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to make International calls from asterisk PBX

[Sam Basan]

International calls are exactly as local phones using the same lines/trunks.

First check your outbound route to verify that your dial plan match your 
dialing international pattern.







Sincerely,



Sam Basan





From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Uzma Anjum
Sent: Monday, February 12, 2018 1:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to make International calls from asterisk PBX



Hello...



I'm running asterisk-13 and international calls are not working in it.How can I 
make it work.Can anyone please tell me.



---
הודעת דוא"ל זו נבדקה לאיתור וירוסים על ידי תוכנת האנטי-וירוס של avast.

https://www.avast.com/antivirus
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to make International calls from asterisk PBX

[Antony Stone]

On Monday 12 February 2018 at 12:25:00, Uzma Anjum wrote:

> Hello...
> 
> I'm running asterisk-13 and international calls are not working in it.How
> can I make it work.Can anyone please tell me.

We are sorry, but all our telepaths and clairvoyants are busy dealing with 
other queries right now.

Please supply us with more information about how you are currently trying to 
place international calls, and what error messages you get in response, and we 
may be able to help you.

Alternatively you may wait for someone to obtain the magical inspiration which 
enables them to diagnose your problem without any details to work from.


Regards,


Antony.

-- 
There are two possible outcomes:

 If the result confirms the hypothesis, then you've made a measurement.
 If the result is contrary to the hypothesis, then you've made a discovery.

 - Enrico Fermi

   Please reply to the list;
 please *don't* CC me.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] how to make International calls from asterisk PBX

[Uzma Anjum]

Hello...

I'm running asterisk-13 and international calls are not working in it.How
can I make it work.Can anyone please tell me.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users