Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?

[gincantalupo]

Hi Alex,

replace with anything which could make Asterisk connect to Skype 
network, make and receive calls, etc...the usual stuff.


Giorgio

On 12/01/2011 02:40 PM, Alex Balashov wrote:

On 12/01/2011 08:30 AM, gincantalupo wrote:


any idea about how to replace Skype For Asterisk?


Replace with what?





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Re: [asterisk-users] Best VoIP conferencing phone ?

[Jamie A. Stapleton]

Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
* 
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html

We have tested all of these in our lab but I would prefer not to be too verbose 
about my preferences on a mailing list.

Please feel free to call me if you want more detail,
-jamie
(804) 412-1601
sip: ja...@cbsiva.com
Skype:  cbsi_jamie



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, November 30, 2011 3:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best VoIP conferencing phone ?

Hi Faisal,

Thanks for reply but I want hardware wase VoIP device. If know please gussed 
me. From google I fould the list of below devices but I am not sure that these 
are best for used or have an issue 

 1)Polycom SoundStation IP 7000

Why it's best: The Polycom SoundStation IP 7000 is the most advanced conference 
phone from the Polycom SoundStation lineup and leaves little to be desired. 
With an amazing 20' 360 radius, the 7000 is perfect for large conference rooms. 
The new HD voice quality (22 kHz) allows.

2) Polycom Voicestation 500

Why it's a best pick: The Polycom VoiceStation 500 is one of the best 
conference phones for a wide variety of reasons. The VoiceStation 500 features 
amazing call quality, 7' 360 radius, Bluetooth connectivity, wired connection, 
background noise reduction, and an attractive design.

3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S

Why it's a best pick: With a 360 10' radius and 8 microphones, everyone is sure 
to be heard with the Panasonic KX-TS730S. The multiple microphones allows for 
everyone sitting in on the conference to be heard uniformly without distortion.

4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone

Why it's a best pick: The Cisco 7937G works via VoIP connection, has stunning 
call clarity, and features a simplistic but expensive design that is easy to 
use. Cisco is an industry leader in IT communication products, and the 7937G is 
no different. The 360 design allows everyone to be heard.

5)Polycom SoundStation VTX 1000

Why it's a best pick: The SoundStation VTX 1000 is an incredible conference 
phone, but it is very pricey and not as good as advertised. The VTX 1000 is 
designed for large conference rooms and features upgradable software (which is 
a huge benefit since the cost is so high), 20' 360 radius.
6)Polycom(r) SoundStation(r) IP 5000
7) GXP2120 6-line Executive HD IP Phone

On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif 
mailto:fai...@vopium.com>> wrote:
I have tried EyeBeam and it worked fine with x members audio conference however 
it need resources (Processing + RAM) per additional line.

Regards,

Faisal Hanif

From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of virendra bhati
Sent: Wednesday, November 30, 2011 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; 
Sam Govind
Subject: [asterisk-users] Best VoIP conferencing phone ?

Hi ,

I know it's might not the right way to asking such stupid question. But I want 
to take help from experts into VoIP fields so I have to decided to post here.

Please help me which will be the best VoIP conferencing phone which will cover 
10 Persians into conferencing with best audio support ?

--

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer


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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer

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Re: [asterisk-users] A new hack?

[C F]

On Thu, Dec 1, 2011 at 8:15 AM, Gordon Henderson
 wrote:
> On Tue, 29 Nov 2011, C F wrote:
>
>> BTW, you were just proven wrong, you need it for this hack.
>
> In addition to the few hundred protected asterisk installations I run, I
> also run a few honeypots.

Protected? You don't know that until the next hack comes out.

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

On Thu, 1 Dec 2011 23:23:56 +0100
Torbjörn Abrahamsson  wrote:

> This was run on an Fedora 8 machine, with perl 5.8.8. I also found it
> odd that the path was not included...
> 
> // T
> 
It seems this is an issue with older versions of perl or at least with
5.8.8. Since this version is used in RHEL/CentOS 5.x that many people
run on their servers, this is a serious problem.

Changing the way tempfile() is called from: 
tempfile("ggl_XX", TMPDIR => 1, UNLINK => 1)
to:
tempfile("ggl_XX", DIR => $tmpdir, UNLINK => 1)
seems to address this issue.

An updated version including this fix can be obtained here:
http://github.com/zaf/asterisk-googletts/tarball/master


Lefteris Zafiris

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Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?

[Hans Witvliet]

On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
> On Thursday 01 December 2011, gincantalupo wrote:
> > Hi all,
> > 
> > any idea about how to replace Skype For Asterisk?
> > 
> > Thank You.
> > 
> > Giorgio
> 
> 1.  Migrate your Skype users over to a better product which supports proper 
> open standards. 
perhaps you missed it, but the installed base of skype is unfortunately
slightly (,,,) larger than the amount of peope that are using a decent
product. Alas


> 2.  Write to your elected representatives asking that they order Skype to 
> release documentation on their protocols to allow third party 
> interoperability  
> (as is already required under EU law). 

3. make it a offence to use any closed source products like skype. >;-)
Huge fines, jail centences or worse.
[How about an appendice to the Thora, Quran or Bible, even better,
forbid it by the sharia]



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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Torbjörn Abrahamsson]

This was run on an Fedora 8 machine, with perl 5.8.8. I also found it odd that 
the path was not included...

// T


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lefteris Zafiris
Sent: den 1 december 2011 22:48
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI script that uses google's text to speech 
engine

On Thu, 1 Dec 2011 21:51:21 +0100
Torbjörn Abrahamsson  wrote:

> This is because you need to add /tmp to the STREAM command, ie:
> 
> print "STREAM FILE /tmp/$tmpname \"$intkey\"\n";
> 
> $tmpname seems to not contain the path, so it will look in
> /var/lib/asterisk/sounds for the file...
> 
> This at least made it work for me... (After fixing some other things
> to make it work with asterisk 1.2...)
> 
> BR,
> Torbjörn Abrahamsson
> 

$tmpname is "supposed" to include the full path together with the temp
dir since its created with the option 'TMPDIR => 1' during the call of
tempfile() and it does so in my system that runs perl 5.14.2. I guess
that might not be true for older versions of perl(?)
Can you please tell me what version of perl you are using?
The oldest perl I can get my hands on is 5.8.8 on RHEL 5.x machines.
I will try to test there and see whats going on.

Thanks for the feedback.


Lefteris Zafiris

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[asterisk-users] Locally bridging channels when using SRTP?

[Jan Blom]

Hello,

I'm trying to setup an Asterisk (version 1.8.8) to do SRTP termination and then 
send the call on to other servers, unencrypted. All the basics work fine.

I want the Asterisk to do as little as possible with the RTP packets and no 
transcoding. We always make sure to force same codec on incoming and outgoing 
call leg.

When not using SRTP, Asterisk does P2P bridging of the RTP packets. That is, 
simply copying the packets, which is the expected result. But when we send in 
SRTP media, Asterisk starts decode/encode voice data instead of just do P2P 
bridging.

I also notice Asterisk doesn't say "Locally bridging channels" in the latter 
case, which might be the clue that we're not doing P2P bridging.

Why can we not use P2P bridging when doing SRTP->RTP media conversion? Is there 
anything we can change in the source code to force packet bridging in this case?


Best regards,
Jan Blom

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

On Thu, 1 Dec 2011 21:51:21 +0100
Torbjörn Abrahamsson  wrote:

> This is because you need to add /tmp to the STREAM command, ie:
> 
> print "STREAM FILE /tmp/$tmpname \"$intkey\"\n";
> 
> $tmpname seems to not contain the path, so it will look in
> /var/lib/asterisk/sounds for the file...
> 
> This at least made it work for me... (After fixing some other things
> to make it work with asterisk 1.2...)
> 
> BR,
> Torbjörn Abrahamsson
> 

$tmpname is "supposed" to include the full path together with the temp
dir since its created with the option 'TMPDIR => 1' during the call of
tempfile() and it does so in my system that runs perl 5.14.2. I guess
that might not be true for older versions of perl(?)
Can you please tell me what version of perl you are using?
The oldest perl I can get my hands on is 5.8.8 on RHEL 5.x machines.
I will try to test there and see whats going on.

Thanks for the feedback.


Lefteris Zafiris

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Re: [asterisk-users] Can't get off Europe/Bucharest timezone

[Nick Khamis]

I'm so sorry, i'm so sorry, i'm so sorry!
Good thing I did not have a chance yet
to transfer it to mysql realtime. It was
in extensions.conf.

Thanks for Everything,

Nick.

On Thu, Dec 1, 2011 at 3:41 PM, Danny Nicholas  wrote:
> Assuming it's nothing quirky in some mysql or odbc, I would do
> - grep "Europe" /etc/asterisk/*
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Thursday, December 01, 2011 2:36 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Can't get off Europe/Bucharest timezone
>
> Hello Everyone,
>
> The timezone is set correctly on the OS America/Toronto:
>
> mv /etc/localtime /etc/localtime.bak
> cp /usr/share/zoneinfo/America/Toronto /etc/localtime
>
> I even tried adding the timezone setting to sip.conf:
>
> timezone=America/Toronto
>
> However. Asterisk wants to be in Bucharest? Thinking about it, I want to be
> in Bucharest!
>
>
> Cheers,
>
> Nick.
>
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>
>
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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Torbjörn Abrahamsson]

This is because you need to add /tmp to the STREAM command, ie:

print "STREAM FILE /tmp/$tmpname \"$intkey\"\n";

$tmpname seems to not contain the path, so it will look in
/var/lib/asterisk/sounds for the file...

This at least made it work for me... (After fixing some other things to make
it work with asterisk 1.2...)

BR,
Torbjörn Abrahamsson



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lefteris
Zafiris
Sent: den 1 december 2011 18:34
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI script that uses google's text to speech
engine

On Thu, 1 Dec 2011 09:43:29 -0500
"bakko"  wrote:

> Hello,
> 
> when I use the Agi, sometimes not play the phrase:
> 
> WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does
> not exist in any format
> 
> Regards

Seems like the script failed to convert the mp3 data that gets from
google to raw slinear. In that case mpg123 or sox
failed to run. It would be very helpful if you could send the full
console output with verbosity set to 3. Please reply to my mail address
so we don't pollute the list.

Thanks for the feedback


Lefteris Zafiris
  


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Re: [asterisk-users] Can't get off Europe/Bucharest timezone

[Danny Nicholas]

Assuming it's nothing quirky in some mysql or odbc, I would do 
- grep "Europe" /etc/asterisk/*


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, December 01, 2011 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Can't get off Europe/Bucharest timezone

Hello Everyone,

The timezone is set correctly on the OS America/Toronto:

mv /etc/localtime /etc/localtime.bak
cp /usr/share/zoneinfo/America/Toronto /etc/localtime

I even tried adding the timezone setting to sip.conf:

timezone=America/Toronto

However. Asterisk wants to be in Bucharest? Thinking about it, I want to be
in Bucharest!


Cheers,

Nick.

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[asterisk-users] Can't get off Europe/Bucharest timezone

[Nick Khamis]

Hello Everyone,

The timezone is set correctly on the OS America/Toronto:

mv /etc/localtime /etc/localtime.bak
cp /usr/share/zoneinfo/America/Toronto /etc/localtime

I even tried adding the timezone setting to sip.conf:

timezone=America/Toronto

However. Asterisk wants to be in Bucharest? Thinking
about it, I want to be in Bucharest!


Cheers,

Nick.

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

On Thu, 1 Dec 2011 11:35:21 -0600
"Danny Nicholas"  wrote:

> I personally don't like the use of mpg123 for playback - would prefer
> use of the internal "Playback/background" functions.  Still seems to
> be a nice effort though.

mpg123 used to convert the mp3 data that we get from google to wav. The
wav file is passed to sox that converts it to raw slinear and then its
played back by asterisk using the 'stream file' agi command.

I don't really like calling all these system commands but I thought it
would be better for the users to have the voice data in sln than mp3
since format_mp3 module isn't available in many installations.


Lefteris Zafiris 

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

On Thu, 01 Dec 2011 17:23:59 +
Kingsley Tart  wrote:

> Hi. Aside from converting spaces to plus signs, you don't encode any
> special characters before putting them in the URL. It might be safer
> to run $line through some sort of encoding before calling Google with
> it, even if most special characters probably don't result in any
> sound. Google say "and" if you give it an ampersand, but unescaped
> you couldn't include that in the string.
> 
> You may decide to have an option to locally cache pre-produced sound
> files in case that phrase is used again.
> 
> Cheers,
> Kingsley.
> 

Thanks for the suggestion. Ther's already some sort of input sanitation:
 $AGI{arg_1} =~ s/[\\\/|*~<>^\(\)\[\]\{\}\n\r]/ /g;
that strips most special characters but i guess it needs some more work.
As for the caching the script supports it already, its enabled by
default and controlled by these 2 variables in the script:
 $usecache   = 1;
 $cachedir   = "/tmp";
Voice data gets stored in the cachedir for future use so we don't have
to fetch it from google each time.


Lefteris Zafiris

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Danny Nicholas]

I personally don't like the use of mpg123 for playback - would prefer use of
the internal "Playback/background" functions.  Still seems to be a nice
effort though.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Thursday, December 01, 2011 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI script that uses google's text to speech
engine

Hi. Aside from converting spaces to plus signs, you don't encode any special
characters before putting them in the URL. It might be safer to run $line
through some sort of encoding before calling Google with it, even if most
special characters probably don't result in any sound.
Google say "and" if you give it an ampersand, but unescaped you couldn't
include that in the string.

You may decide to have an option to locally cache pre-produced sound files
in case that phrase is used again.

Cheers,
Kingsley.

On Thu, 2011-12-01 at 02:42 +0200, Lefteris Zafiris wrote:
> Hello,
> I have written an AGI script for asterisk that uses google translate 
> for text to speech synthesis.
> It supports a variety of different languages, local caching for the 
> voice data and wideband audio.
> The voice in most languages is female and the quality of the 
> synthesized speech is very high.
> More info about the script can be found here:
> http://zaf.github.com/asterisk-googletts/
> the first public release ca be obtained here:
> https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts
> -0.2.tar.gz
> 
> To get a sample of the speech synthesis quality try this link:
> http://translate.google.com/translate_tts?tl=en&q=this+is+a+test+for
> +google+text+to+speech+engine
> 
> The code is still very young so suggestions, comments and bug reports 
> are more than welcome.
> 
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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

On Thu, 1 Dec 2011 09:43:29 -0500
"bakko"  wrote:

> Hello,
> 
> when I use the Agi, sometimes not play the phrase:
> 
> WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does
> not exist in any format
> 
> Regards

Seems like the script failed to convert the mp3 data that gets from
google to raw slinear. In that case mpg123 or sox
failed to run. It would be very helpful if you could send the full
console output with verbosity set to 3. Please reply to my mail address
so we don't pollute the list.

Thanks for the feedback


Lefteris Zafiris
  


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Re: [asterisk-users] AGI script that uses google's text to speech engine

[Kingsley Tart]

Hi. Aside from converting spaces to plus signs, you don't encode any
special characters before putting them in the URL. It might be safer to
run $line through some sort of encoding before calling Google with it,
even if most special characters probably don't result in any sound.
Google say "and" if you give it an ampersand, but unescaped you couldn't
include that in the string.

You may decide to have an option to locally cache pre-produced sound
files in case that phrase is used again.

Cheers,
Kingsley.

On Thu, 2011-12-01 at 02:42 +0200, Lefteris Zafiris wrote:
> Hello,
> I have written an AGI script for asterisk that uses google translate
> for text to speech synthesis.
> It supports a variety of different languages, local caching for the
> voice data and wideband audio.
> The voice in most languages is female and the quality of the
> synthesized speech is very high.
> More info about the script can be found here:
> http://zaf.github.com/asterisk-googletts/
> the first public release ca be obtained here:
> https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts-0.2.tar.gz
> 
> To get a sample of the speech synthesis quality try this link:
> http://translate.google.com/translate_tts?tl=en&q=this+is+a+test+for
> +google+text+to+speech+engine
> 
> The code is still very young so suggestions, comments and bug reports
> are more than welcome.
> 
> --
> Lefteris Zafiris
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Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?

[Tom Browning]

On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
 wrote:
> Hi all,
>
> any idea about how to replace Skype For Asterisk?
>
> Thank You.
>
> Giorgio
>

We are going through this right now and have chosen to "Pay The Man"
via per channel subscription to Skype Connect.

Watch the fun video at:
http://www.skype.com/intl/en/business/skype-connect/   :-)

Skype-For-Asterisk is a vastly superior product/service but someone at
Skype woke up one day and said, "Hey we can't let that product
succeed and lose control of some valuable fees".

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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar [SOLVED]

[Jimmy Godbout]

From experience, that model is not reliable. I have changed those with HP 
Procurve and my problems were gone.

Just my 0.02

Jimmy

> -Original Message-
> From: oza_4...@yahoo.fr
> Sent: Thu, 1 Dec 2011 13:44:44 +0100
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
> [SOLVED]
> 
> 2011/11/30 Mike 
> 
>> Hi Olivier,
>> 
>> ** **
>> 
>> It if occurs only on the sidecar, I would imagine this is either a
>> defective sidecar/Polycom phone, or a defective PoE switch not giving
>> enough power. Changing PoE port would eliminate of confirm the PoE port
>> being the issue, but I’m betting on a Polycom defect.
>> 
>> ** **
>> 
>> Make sure the PoE port is configured (if it`s a smart switch) to send
>> maximum power to the port, with a sidecar I think the phone requires
>> 12W.
>> 
>> 
>> ** **
>> 
>> Regards,
>> 
>> ** **
>> 
>> Mike
>> 
>> 
>> 
> Hello,
> 
> As suggested here, we tried this morning, to check if the problem could
> come from PoE.
> 
> When my collegue arrived there, he had the chance to see the sidecar in
> its
> "broken mode": names missing on the LCD screen and so on.
> Plugging a power supplying instantly solved the issue (no need to reboot
> anything).
> 
> I still can explain myself why a PoE switch (a Linksys SRW224P) would
> succeed or fail to deliver power to a plugged IP phone, given that only a
> couple of Polycom phones are using this switch a power source.
> 
> Thanks for everyone.
> Cheers


FREE 3D EARTH SCREENSAVER - Watch the Earth right on your desktop!
Check it out at http://www.inbox.com/earth



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Re: [asterisk-users] A new hack?

[Tom Browning]

On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson
 wrote:

> Yes, I know exactly how Fail2Ban works.

Then you should be able to proffer a better argument of why it isn't necessary.

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Re: [asterisk-users] AGI script that uses google's text to speech engine

[bakko]

Hello,

when I use the Agi, sometimes not play the phrase:

WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does not exist 
in any format

Regards
  - Original Message - 
  From: Lefteris Zafiris 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, November 30, 2011 7:42 PM
  Subject: [asterisk-users] AGI script that uses google's text to speech engine


  Hello,
  I have written an AGI script for asterisk that uses google translate for text 
to speech synthesis.
  It supports a variety of different languages, local caching for the voice 
data and wideband audio.
  The voice in most languages is female and the quality of the synthesized 
speech is very high.
  More info about the script can be found here: 
http://zaf.github.com/asterisk-googletts/
  the first public release ca be obtained here: 
https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts-0.2.tar.gz

  To get a sample of the speech synthesis quality try this link:
  
http://translate.google.com/translate_tts?tl=en&q=this+is+a+test+for+google+text+to+speech+engine

  The code is still very young so suggestions, comments and bug reports are 
more than welcome.

  --
  Lefteris Zafiris



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Re: [asterisk-users] Sound files with MixMonitor not playable with Media Player

[Steve Edwards]

On Thu, 1 Dec 2011, Jonas Kellens wrote:

Like I said : I can play the sound file with Totem on Linux or 
VLC-player on Windows. So it's not that the wav-file has no sound...


Can you post a link to a sample file?

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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?

[A J Stiles]

On Thursday 01 December 2011, gincantalupo wrote:
> Hi all,
> 
> any idea about how to replace Skype For Asterisk?
> 
> Thank You.
> 
> Giorgio

1.  Migrate your Skype users over to a better product which supports proper 
open standards.  

2.  Write to your elected representatives asking that they order Skype to 
release documentation on their protocols to allow third party interoperability  
(as is already required under EU law). 

-- 
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Answers come *after* questions.

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Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?

[Alex Balashov]

On 12/01/2011 08:30 AM, gincantalupo wrote:


any idea about how to replace Skype For Asterisk?


Replace with what?


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[asterisk-users] Skype For Asterisk (SFA): any replacement?

[gincantalupo]

Hi all,

any idea about how to replace Skype For Asterisk?

Thank You.

Giorgio

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Re: [asterisk-users] A new hack?

[Gordon Henderson]

On Tue, 29 Nov 2011, C F wrote:


BTW, you were just proven wrong, you need it for this hack.


In addition to the few hundred protected asterisk installations I run, I 
also run a few honeypots.


Gordon

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Re: [asterisk-users] A new hack?

[Gordon Henderson]

On Wed, 30 Nov 2011, jon pounder wrote:


On 11/30/2011 09:01 AM, Tom Browning wrote:

I agree - its a bad comparison of 2 different things meant for different 
purposes.


iptables is enforcement, fail2ban is detection.


iptables can also detect and log these detections.

if you have time to sit and make up iptables rules by hand during every hack 
attempt


I don't.

Gordon

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Re: [asterisk-users] A new hack?

[Gordon Henderson]

On Wed, 30 Nov 2011, Tom Browning wrote:


On Tue, Nov 29, 2011 at 4:44 PM, john Millican  wrote:


Maybe I am misunderstanding the gist of the comment


OP offered an invalid comparison of how iptables is better than Fail2Ban.

Whether or not OP knew that Fail2Ban simply feeds rules to iptables is
unclear from his comments.


Yes, I know exactly how Fail2Ban works.

Gordon

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Re: [asterisk-users] A new hack?

[Gordon Henderson]

On Tue, 29 Nov 2011, C F wrote:


On Mon, Nov 28, 2011 at 10:57 AM, Tom Browning  wrote:

On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
 wrote:

Linux has excellent built-in subsystems to control firewalling and so on
without resorting to external programs. It's called iptables. If you know
how to use them, then using an external resource such as fail2ban is
unneccessary.


That's like saying you don't need FreePBX because you have this thing
called Asterisk.


Very well put.


Indeed. I don't need (nor use) FreePBX.

Gordon

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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar [SOLVED]

[Administrator TOOTAI]

Le 01/12/2011 13:44, Olivier a écrit :

[...]
I still can explain myself why a PoE switch (a Linksys SRW224P) would 
succeed or fail to deliver power to a plugged IP phone, given that 
only a couple of Polycom phones are using this switch a power source.
I think your switch deliver a max value of power per port, the phone and 
side-car take are just on this limit.


--
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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar [SOLVED]

[Olivier]

2011/11/30 Mike 

> Hi Olivier,
>
> ** **
>
> It if occurs only on the sidecar, I would imagine this is either a
> defective sidecar/Polycom phone, or a defective PoE switch not giving
> enough power. Changing PoE port would eliminate of confirm the PoE port
> being the issue, but I’m betting on a Polycom defect.
>
> ** **
>
> Make sure the PoE port is configured (if it`s a smart switch) to send
> maximum power to the port, with a sidecar I think the phone requires 12W.
> 
>
> ** **
>
> Regards,
>
> ** **
>
> Mike
>
>
>
Hello,

As suggested here, we tried this morning, to check if the problem could
come from PoE.

When my collegue arrived there, he had the chance to see the sidecar in its
"broken mode": names missing on the LCD screen and so on.
Plugging a power supplying instantly solved the issue (no need to reboot
anything).

I still can explain myself why a PoE switch (a Linksys SRW224P) would
succeed or fail to deliver power to a plugged IP phone, given that only a
couple of Polycom phones are using this switch a power source.

Thanks for everyone.
Cheers
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[asterisk-users] Populate CDR issues

[Harel Cohen]

Hello list,
I'm trying to populate my CDR logs with values which are available after the 
call has started (e.g. signalling IP of remote user, media IP, codec etc.). 
While CHANNEL function give me all I need for the incoming leg (leg A), I can't 
get the relevant values for the outgoing channel. I've tried using the option 
'U' with my dial command (execute subroutine for called channel after called 
channel answered but before the call is bridged). While this throws the correct 
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC 
and the table therein contains the relevant fields.

This is the console with 'very-verbose' output for the 'Dial' application where 
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP 
192.168.20.226. My comments added prefixed by ** and on separate line:

** channel here is source channel: SIP/office_Admin2-0015
[Dec  1 12:14:31] -- Executing [316@InternalDP:5] 
Dial("SIP/office_Admin2-0015", "SIP/office_ServerRoom,,FgU(jump2SetVar)") 
in new stack
[Dec  1 12:14:31]   == Using UDPTL CoS mark 5
[Dec  1 12:14:31]   == Using SIP RTP CoS mark 5
[Dec  1 12:14:31] -- Called SIP/office_ServerRoom
[Dec  1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec  1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec  1 12:14:33] -- SIP/office_ServerRoom-0016 answered 
SIP/office_Admin2-0015
** from here the channel is the destination channel: 
SIP/office_ServerRoom-0016
[Dec  1 12:14:33] -- Executing [s@jump2SetVar:1] 
Gosub("SIP/office_ServerRoom-0016", "SetVar,postdial,1") in new stack
** This is how I obtain channel information:
** exten => 
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec  1 12:14:33] -- Executing [postdial@SetVar:1] 
Set("SIP/office_ServerRoom-0016", "CDR(chanoutsigip)=192.168.20.226:5065") 
in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:2] 
Set("SIP/office_ServerRoom-0016", 
"CDR(chanoutmediaip)=192.168.20.226:23008") in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:3] 
Set("SIP/office_ServerRoom-0016", "CDR(chanoutcodec)=g729") in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:4] 
Goto("SIP/office_ServerRoom-0016", "endsub,1") in new stack
[Dec  1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec  1 12:14:33] -- Executing [endsub@SetVar:1] 
Return("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Executing [s@jump2SetVar:2] 
Return("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1] 
NoOp("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Auto fallthrough, channel 
'SIP/office_ServerRoom-0016' status is 'UNKNOWN'
[Dec  1 12:14:33] -- Remotely bridging SIP/office_Admin2-0015 and 
SIP/office_ServerRoom-0016

When call is terminated the relevant fields in the database for 
CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with 
their default values (typically blank or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with 
info from called channel (leg B)?

Thank you for your replies...

Harel



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Re: [asterisk-users] hwo to stok variable wiith menu

[salaheddine elharit]

Hi Noll,

all works perfectly thanks a lot for your help and support i really
appreciate it :)

Best Regards

2011/12/1 Dale Noll 

>
> On 11/30/2011 11:13 AM, salaheddine elharit wrote:
>
>> i have last question regarding this thread
>> with exten => 3,n,MYSQL(Query resultid ${connid} insert into test (
>> option_name ) values ('${CALLERID(num)}'))
>> i can store the phone number without issue
>> i need also the date and hour fo call in the "count coulum"
>> could you please give me the syntex
>> best regards
>>
>>
> The example table that I gave originally was before I knew what you were
> looking to do. I assumed, incorrectly that you simply wanted to track how
> many times an option was selected in the menu.
> I would recommend that you create a table specifically for this
> application.
>
> That table may look like this.  Please name the table and columns
> appropriately for your application.
>
> create table option_three (
> calldatedatetime,
> calleridvarchar(40)
> )
>
> Then the sql would look something like this...
>  exten => 3,n,MYSQL(Query resultid ${connid} insert into option_three (
> calldate, callerid ) values ( now(), '${CALLERID(num)}'))
>
>
> Dale
>
> --
> "The truth speaks for itself. I'm just the messenger."
> Lyta Alexander - Babylon 5
>
>
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Re: [asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)

[James Mutuku]

Thanks for Carlos for the response,

I have worked with bare asterisk + freepbx before. the mypbx was just
an example but my reference to  appliances as a whole.

The appliances seem to have lower entry costs.



On 12/1/11, Carlos Alvarez  wrote:
> At the most basic level, typically an appliance will have a GUI and be
> geared towards non-tech installation.  Loading bare Asterisk on a server is
> very different.  Do you want a GUI or bare Asterisk?
>
> BTW, the MyPBX product is not a Digium product, it's from an oriental
> company named Yeastar.  My experience in talking to them about their phones
> has been so-so.  Historically we've had awful experiences with other
> Chinese phone vendors and have stopped considering products from Chinese
> companies.  We did not actually try Yeastar products.
>
>
> On Wed, Nov 30, 2011 at 3:39 PM, James Mutuku  wrote:
>
>> Hi,
>>
>> I am looking into advising a client on the pro's and cons of using
>> Installing asterisk on a server vs appliance(e.g digium mypbx).  the
>> appliance seems cheaper initially.
>>
>> From experience,  what would be pro and cons for either option?
>>
>> --
>> Best Regards,
>> James Mutuku Ndeti
>> Agile Systems Limited
>> +254722490994
>> www.agile.co.ke
>> www.zetu.co.ke
>>
>> Has your organization implemented a customer relationship management
>> (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our
>> CRM can help you achieve better customer satisfaction and sales
>>
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>>
>
>
>
> --
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> TelEvolve
> 602-889-3003
>


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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

[Olivier]

2011/11/30 Marco Mooijekind 

> Maybe use a power supply instead of PoE, see if problem still occurs.
> Marco.
>
Yes, that's what I meant by "not using PoE".

Op 30 nov. 2011 18:46 schreef "Olivier"  het volgende:
>
>
>>
>> 2011/11/30 Mike 
>>
>>> Hi Olivier,
>>>
>>> ** **
>>>
>>> It if occurs only on the sidecar, I would imagine this is either a
>>> defective sidecar/Polycom phone, or a defective PoE switch not giving
>>> enough power. Changing PoE port would eliminate of confirm the PoE port
>>> being the issue, but I’m betting on a Polycom defect.
>>>
>>> ** **
>>>
>>> Make sure the PoE port is configured (if it`s a smart switch) to send
>>> maximum power to the port, with a sidecar I think the phone requires 12W.
>>>
>> This info is very interesting.
>> I wouldn't be too surprised that a PoE switch not supplying its theorical
>> 15W output on a long period.
>> I'll try to use work around this possible cause by not using PoE.
>>
>> In any case, I'll report my findings here.
>>
>>> 
>>>
>>> ** **
>>>
>>> Regards,
>>>
>>> ** **
>>>
>>> Mike
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
>>> *Sent:* Wednesday, November 30, 2011 10:27 AM
>>>
>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>> *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar**
>>> **
>>>
>>> ** **
>>>
>>> Hello,
>>>
>>>
>>> On one location, I've got from time to time (let say one a week) the
>>> following issue :
>>> the phone SoundPoint 650 works ok (can call or answer, display and sound
>>> are ok),
>>> the sidecar looses its display : entries on sidecar's LCD screen are not
>>> displayed anymore, or names are truncated, or BLF are not shown or updated.
>>>
>>> I only have one SPIP650 on this system so I can't compare with others.
>>>
>>> What could be the root cause of this ?
>>>
>>> Regards
>>>
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>>
>>
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>
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Re: [asterisk-users] s/n ratio detection etc...

[Yasin SULUHAN]

On Wed, Nov 30, 2011 at 4:48 PM, Danny Nicholas  wrote:

> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN
> *Sent:* Wednesday, November 30, 2011 8:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] s/n ratio detection etc...
>
> ** **
>
> ** **
>
> On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas  wrote:
> 
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN
> *Sent:* Wednesday, November 30, 2011 6:25 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] s/n ratio detection etc...
>
>  
>
> Hi everybody,
>
> I' ve been following this list for a while now.
>
> Is there a way to detect the individual and cumulative s/n ratio values
> for the incoming calls in Asterisk or any other Call Center solution?...**
> **
>
>  
>
> Either I need to finish my coffee or this should be worded better:
>
>
> Sorry about this. This request just came in from a client and we need an
> answer very quickly.
>  
>
> Is there a way the detect the individual and cumulative signal-to-noise
> ratio values for incoming calls to Asterisk (or any other Call Center
> solution)?
>
>  
>
>  
>
> This depends on
>
> 1.   How are the calls delivered to Asterisk (we will ignore the
> “other call center” since this is an Asterisk discussion board)?
> SIP/DAHDI(PSTN/PRI/E1/ETC)?
>
> DAHDI
>  
>
> 2.   What version of Asterisk?
>
> 1.8.7
>  
>
> 3.   Do you want “built-in” methods or could other methods such as
> daemons be used?  
>
> either way would be ok.
>
> 
>
> Your best bet as I understand it would be to use dahdi_tools to monitor
> your lines or to use mixmonitor to record the calls so you can review and
> tune problems as needed.   Either of these options would cost you some
> overhead in processor usage and disk space.
>
>
>
Again, thank you for your help... Much appreciated...


>
> Thank you for your quick response
>



>
> 
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Re: [asterisk-users] how to find out one way latency

[Hans Witvliet]

On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote:

> 
> You can make a pretty good prediction with ping.
> "sudo ping -f -i .02 -s 180 -Q 0xb8 [ip]" gives a tolerable simulation 
> of voip traffic.  let it run for awhile, then press ctrl+c and see how 
> many packets were dropped and also check the mdev number.  If mdev is 
> low and packet loss is almost nothing then you can expect decent voice 
> quality.  It may not be a 100% perfect test, but I'll bet you a vast 
> majority of the time I can do that test and tell you whether it's going 
> to suck.
> 
> latency by itself with low jitter and no packet loss just means delay.  
> It's a matter of opinion and circumstance how tolerable delay is, but I 
> think your 230ms ping is at the upper edge of what most people can live 
> with.  Much more than that and you'll be tempted to say 'over' at the 
> end of sentence.
> 
> --
Fully agree,

Actually, you can do better than just a ping, but it takes some time,
equipment and experience:

What you can do, is adding an extra box inbetween your voip-client and
voip-server, and introduce all kinds of "real-life" circumstances.
I mean artificial delay, loss, resequencing, duplicating packages,
reduced bandwith. We've done it some time ago as an "satelite simulator"
You can build it aroud any *bsd/linux box with multiple nics.

The basic idea's you can find at http://lartc.org/
If you combine it with the echo function from asterisk, you can decide
for yourself what it acceptable and what not.

For one of my projects i push the echo destination as the "default" sip
connection to their soft phone, as i noticed that people at the other
side of town regularly have a worse connection then people using umts or
satelite. Main culprit (in my case) is ill-configured WIFI-setup.
Latencies of over 10,000 ms and loss of 80% are daily events.
And people complaining


hw


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