Re: [asterisk-users] Many tests from TestSuite fail with "Asterisk 127.0.0.1 received error: FRACK!"

2017-10-31 Thread George Joseph
On Tue, Oct 31, 2017 at 7:02 AM, Olivier  wrote:

> Hello,
>
> Thanks to Tzafrir and George help, I could run Asterisk TestSuite for the
> first time on a fresh Debian Stretch setup.
> TestSuite is installed with "apt-get install asterisk-testsuite" and
> Asterisk itself is stopped (with "service asterisk stop") before executing
> (as root) asterisk-tests-run.
>
> On console, I can see 14 instances of the following error
>
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
>


Can you post the results from a single test?


>
> Is it worth reporting this ?
>
> 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
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Remote Phonebook with Thomson ST2022

2017-10-31 Thread Ludovic Gasc
Welcome to the world of random HTTP client support of hardphones ;-)
I have a lot of funny stories to tell about that.

My advices:

   1. Upgrade to the latest firmware
   2. Sniff with tcpdump / tshark
   3. Check the content of the capture and compare with a first class
   citizen HTTP client like curl, Chrome...
   4. If nothing obvious appears, contact their support with this pcap
   trace.

Regards.
--
Ludovic Gasc (GMLudo)

2017-10-25 19:45 GMT+02:00 Luca Bertoncello :

> Hi list!
>
> I'm a little bit OT, I know, but maybe someone here can help me...
>
> I have a Thomson ST2022 and I'd like to have the phonebook on a Server,
> synchronized with my DAV phonebook.
>
> I created a phonebook.xml so:
>
> 
> 
>   
> Test number
> 12345678
>   
> 
>
> Then I set http://www.myserver.de/phonebook.xml in the configuration for
> the
> phonebook.
>
> As soon as I search an entry, the phone crashes... :(
> In the Apache-Log I see very curiously this:
>
> 192.168.200.10 - - [25/Oct/2017:19:38:40 +0200] "GET
> http://www.myserver.de/phonebook.xml HTTP/1.1" 200 36611
>
> instead of:
>
> 192.168.200.10 - - [25/Oct/2017:19:38:40 +0200] "GET /phonebook.xml
> HTTP/1.1" 200 36611
>
> Can someone help me?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Remote Phonebook with Thomson ST2022

2017-10-31 Thread Olivier
Hi,

Have you read about ST2030 remote phonebook examples ?
As both phones share the same engineering and ST2030 was quite successfull,
years ago, that may be worth looking at ST2030 documentation.

Best regards

2017-10-25 19:45 GMT+02:00 Luca Bertoncello :

> Hi list!
>
> I'm a little bit OT, I know, but maybe someone here can help me...
>
> I have a Thomson ST2022 and I'd like to have the phonebook on a Server,
> synchronized with my DAV phonebook.
>
> I created a phonebook.xml so:
>
> 
> 
>   
> Test number
> 12345678
>   
> 
>
> Then I set http://www.myserver.de/phonebook.xml in the configuration for
> the
> phonebook.
>
> As soon as I search an entry, the phone crashes... :(
> In the Apache-Log I see very curiously this:
>
> 192.168.200.10 - - [25/Oct/2017:19:38:40 +0200] "GET
> http://www.myserver.de/phonebook.xml HTTP/1.1" 200 36611
>
> instead of:
>
> 192.168.200.10 - - [25/Oct/2017:19:38:40 +0200] "GET /phonebook.xml
> HTTP/1.1" 200 36611
>
> Can someone help me?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Measuring total end-to-end latency

2017-10-31 Thread Olivier
Hi,

I don't have a direct answer, but I've read several times about purposely
customized system over the PSTN, echoing incoming incoming audio to produce
metrics
when troubleshooting call quality.

I alse remember a thing called Recqual targeting the same goal.

I hope this helps

2017-10-31 11:53 GMT+01:00 Antony Stone <
antony.st...@asterisk.open.source.it>:

> Hi.
>
> Does anyone have some recommendations for measuring total end-to-end
> latency
> (by which I mean: the time between person A saying something and person B
> hearing it) when there are both SIP and PSTN/analogue/mobile legs in the
> call
> path?
>
> Examples:
>
> Person A has a SIP phone registered to Asterisk, which has a SIP trunk to a
> connectivity provider, who has connections to PSTN (analogue landline)
> connectivity providers and to mobile network (Vodafone, Orange, etc)
> providers.
>
> Person B might answer the call on an analogue landline telephone.
>
> Person C might answer the call on a mobile phone (perhaps on its home
> network,
> perhaps roaming on a foreign network).
>
>
> Is there any way to measure total latency of calls between A and B or A
> and C?
>
>
> Thanks in advance for any ideas / suggestions.
>
>
> Antony.
>
> --
> "Remember: the S in IoT stands for Security."
>
>  - Jan-Piet Mens
>
>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
>
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[asterisk-users] Many tests from TestSuite fail with "Asterisk 127.0.0.1 received error: FRACK!"

2017-10-31 Thread Olivier
Hello,

Thanks to Tzafrir and George help, I could run Asterisk TestSuite for the
first time on a fresh Debian Stretch setup.
TestSuite is installed with "apt-get install asterisk-testsuite" and
Asterisk itself is stopped (with "service asterisk stop") before executing
(as root) asterisk-tests-run.

On console, I can see 14 instances of the following error

asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!, Failed
assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of astobj2.c
asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!, Failed
assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of astobj2.c
asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!, Failed
assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of astobj2.c

Is it worth reporting this ?

Best regards
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Re: [asterisk-users] How tu run runtests.py on Debian Stretch ? [SOLVED]

2017-10-31 Thread Olivier
2017-10-31 12:14 GMT+01:00 Tzafrir Cohen :

> On Tue, Oct 31, 2017 at 12:07:57PM +0100, Olivier wrote:
> > Hello,
> >
> > I'm giving asterisk-testsuite package a try on a fresh Debian Stretch
> setup.
> >
> > I've got this:
> > # /usr/share/asterisk-testsuite/runtests.py
> > Traceback (most recent call last):
> >   File "/usr/share/asterisk-testsuite/runtests.py", line 24, in 
> > from asterisk.version import AsteriskVersion
> > ImportError: No module named asterisk.version
> >
> > 1. Which documentation shall I look for ?
> > I've found [1]
> >
> >
> > 2. How shall I run runtests.py ? As asterisk user ? As root ?
>
> asterisk-tests-run
>

Using asterisk-tests-run runs OK !
Thank you very much for helping.


>
> That said, that package needs more work.
>

Probably and diving into it is intimidating given how large is TestSuite
scope.

>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
> --
> _
> -- 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
>
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Re: [asterisk-users] How tu run runtests.py on Debian Stretch ?

2017-10-31 Thread George Joseph
On Tue, Oct 31, 2017 at 5:07 AM, Olivier  wrote:

> Hello,
>
> I'm giving asterisk-testsuite package a try on a fresh Debian Stretch
> setup.
>
> I've got this:
> # /usr/share/asterisk-testsuite/runtests.py
> Traceback (most recent call last):
>   File "/usr/share/asterisk-testsuite/runtests.py", line 24, in 
> from asterisk.version import AsteriskVersion
> ImportError: No module named asterisk.version
>

What version of python are you using?

Try
PYTHONPATH=./lib/python ./runtests.py
and see if that makes a difference


>
> 1. Which documentation shall I look for ?
> I've found [1]
>


That and
https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite


>
>
> 2. How shall I run runtests.py ? As asterisk user ? As root ?
>

Most tests can be run as a normal user.  A few that use the pcap library
need root.  They should be ones that require yappcap.


>
>
> Best
>
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Running+the+
> Asterisk+Test+Suite
>
> --
> _
> -- 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
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] How tu run runtests.py on Debian Stretch ?

2017-10-31 Thread Tzafrir Cohen
On Tue, Oct 31, 2017 at 12:07:57PM +0100, Olivier wrote:
> Hello,
> 
> I'm giving asterisk-testsuite package a try on a fresh Debian Stretch setup.
> 
> I've got this:
> # /usr/share/asterisk-testsuite/runtests.py
> Traceback (most recent call last):
>   File "/usr/share/asterisk-testsuite/runtests.py", line 24, in 
> from asterisk.version import AsteriskVersion
> ImportError: No module named asterisk.version
> 
> 1. Which documentation shall I look for ?
> I've found [1]
> 
> 
> 2. How shall I run runtests.py ? As asterisk user ? As root ?

asterisk-tests-run

That said, that package needs more work.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

-- 
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[asterisk-users] Measuring total end-to-end latency

2017-10-31 Thread Antony Stone
Hi.

Does anyone have some recommendations for measuring total end-to-end latency 
(by which I mean: the time between person A saying something and person B 
hearing it) when there are both SIP and PSTN/analogue/mobile legs in the call 
path?

Examples:

Person A has a SIP phone registered to Asterisk, which has a SIP trunk to a 
connectivity provider, who has connections to PSTN (analogue landline) 
connectivity providers and to mobile network (Vodafone, Orange, etc) 
providers.

Person B might answer the call on an analogue landline telephone.

Person C might answer the call on a mobile phone (perhaps on its home network, 
perhaps roaming on a foreign network).


Is there any way to measure total latency of calls between A and B or A and C?


Thanks in advance for any ideas / suggestions.


Antony.

-- 
"Remember: the S in IoT stands for Security."

 - Jan-Piet Mens

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

-- 
_
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[asterisk-users] How tu run runtests.py on Debian Stretch ?

2017-10-31 Thread Olivier
Hello,

I'm giving asterisk-testsuite package a try on a fresh Debian Stretch setup.

I've got this:
# /usr/share/asterisk-testsuite/runtests.py
Traceback (most recent call last):
  File "/usr/share/asterisk-testsuite/runtests.py", line 24, in 
from asterisk.version import AsteriskVersion
ImportError: No module named asterisk.version

1. Which documentation shall I look for ?
I've found [1]


2. How shall I run runtests.py ? As asterisk user ? As root ?


Best


[1]
https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite
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