date:20110428


On 28/04/11 8:00 PM, virendra bhati wrote:

Hi,

Asterisk server is running on this machine then I tested and I got this
message after run the script.


What user are you running the script as?

It looks like you're running it as a web server when Asterisk is running 
as root?


Try running the script from the commandline while logged in as root to 
confirm:


php script.php

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Re: [asterisk-users] how to know status of asterisk from php

2011-04-28 Thread virendra bhati

Hi,

I am running this script from wabsite and want to make it just like FreePBX
show status information of  Asterisk, mysql etc.

*Asterisk is running as root* at my end.

When I start script from command prompt then I am getting error message..

[root@cent68 mtnl]# php temp.php
PHP Parse error:  syntax error, unexpected ',' in
/var/www/html/mtnl/temp.php on line 3

Parse error: syntax error, unexpected ',' in /var/www/html/mtnl/temp.php on
line 3



On Thu, Apr 28, 2011 at 3:03 PM, Matt Riddell li...@venturevoip.com wrote:

 On 28/04/11 8:00 PM, virendra bhati wrote:

 Hi,

 Asterisk server is running on this machine then I tested and I got this
 message after run the script.


 What user are you running the script as?

 It looks like you're running it as a web server when Asterisk is running as
 root?

 Try running the script from the commandline while logged in as root to
 confirm:

 php script.php

 --
 Cheers,

 Matt Riddell
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/exchange.php (Full ITSP Solution)
 http://www.venturevoip.com/cc.php (Call Centre Solutions)


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Thanks and regards

 Virendra Bhati
+91-9172341457
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Danny Nicholas

This may be Gas on the fire, but I think somebody (Digium/the
community/etc) needs to make a 1.4 parallel installation of 1.8 and get
the baseline in order.  Once the parallel features are functional, then we
can all sweat the problems in the extra features.  If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish.  BTW, where does 1.10 fit into this
conversation?



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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Bryant Zimmerman

I will throw in my 2 cents on this. I agree that 1.8 is not as stable as it 
needs to be. From my perspective I will continue to use the 1.4.x or 
1.6.2.x release that is the best fit for me and it should continue to do 
what it does and it get's it's security releases.

If the primary development focus is moved to 1.8 to get the lead out and 
stabilize it than that is what I want. New work on 1.10 should only be 
under taken after 1.8.x is stable then we can tinker with the newer stuff. 
Making it stable makes it stronger. As far as I can see 1.4.x is stable and 
that is what people want use it until 1.8.x is where you want it but test 
1.8.x help find the bugs so you can make the move otherwise stay with the 
solid 1.4.x and wait for others to find the bugs in the newer versions. I 
know of several companies that are on 1.2 and will make the move to a new 
version only if 1.2 fails them and it has not for their needs. Again we do 
need 1.8 to be stabilized quickly the stuck voicemail issues and system 
crashes are driving me crazy. 

Thanks to all of the developers who work on asterisk. The core makes my 
business possible. Keep up the good work

Thanks
Bryant

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Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?


Thanks for rely,

Actually i have build OSLEC with the help of 
http://www.rowetel.com/blog/?page_id=454 

And now i am getting error at loading module 

root@shirley:~# lsmod | grep echo
dahdi_echocan_mg2   5662  23
dahdi 210313  50 dahdi_echocan_mg2,wanpipe
echo5253  0
root@shirley:~# modprobe dahdi_echocan_oslec
FATAL: Error inserting dahdi_echocan_oslec 
(/lib/modules/2.6.32-30-preempt/dahdi/dahdi_echocan_oslec.ko): Unknown symbol 
in module, or unknown parameter (see dmesg)


What do you suggest ?

-S

Date: Thu, 28 Apr 2011 08:54:30 +0100
From: tiago.ge...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?

linux-dahdi/README has a section on how to compile and install oslec

On 27 April 2011 22:15, Anthony Messina amess...@messinet.com wrote:

On 04/27/2011 02:06 PM, satish patel wrote:

 Which echo cancellation is good between OSLEC and MG2. Dahdi by default use 
 MG2 echo cancellation on channel.  If i want to use OSLEC then what should i 
 need to do ? Do i need to recompile dahdi with OSLEC ?



Yes, you would need to compile the OSLEC kernel module.  Or, if you are

using a RedHat/Fedora based distro, you're welcome to use the

dahdi-linux and dahdi-linux-kmod RPMS I build here.  I include OSLEC

with the dahdi-linux-kmod build.



http://messinet.com/rpms/



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Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?


AH! i dig into kernel and i found there was already echo.ko module exist in 
kernel  

 ls -l /lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko
-rw-r--r-- 1 root root 10840 2011-03-01 21:48 
/lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko

I remove that echo.ko module and install dahdi echo.ko module and it allowed me 
to load oslec module 

I believe i should download kernel-2.6.32.30 and build oslec again that source 
right ? 

-S


From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 28 Apr 2011 14:35:25 +
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?








Thanks for rely,

Actually i have build OSLEC with the help of 
http://www.rowetel.com/blog/?page_id=454 

And now i am getting error at loading module 

root@shirley:~# lsmod | grep echo
dahdi_echocan_mg2   5662  23
dahdi 210313  50 dahdi_echocan_mg2,wanpipe
echo5253  0
root@shirley:~# modprobe dahdi_echocan_oslec
FATAL: Error inserting dahdi_echocan_oslec 
(/lib/modules/2.6.32-30-preempt/dahdi/dahdi_echocan_oslec.ko): Unknown symbol 
in module, or unknown parameter (see dmesg)


What do you suggest ?

-S

Date: Thu, 28 Apr 2011 08:54:30 +0100
From: tiago.ge...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?

linux-dahdi/README has a section on how to compile and install oslec

On 27 April 2011 22:15, Anthony Messina amess...@messinet.com wrote:

On 04/27/2011 02:06 PM, satish patel wrote:

 Which echo cancellation is good between OSLEC and MG2. Dahdi by default use 
 MG2 echo cancellation on channel.  If i want to use OSLEC then what should i 
 need to do ? Do i need to recompile dahdi with OSLEC ?



Yes, you would need to compile the OSLEC kernel module.  Or, if you are

using a RedHat/Fedora based distro, you're welcome to use the

dahdi-linux and dahdi-linux-kmod RPMS I build here.  I include OSLEC

with the dahdi-linux-kmod build.



http://messinet.com/rpms/



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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E




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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 PS. Please don't start a discussion about 1.8 quality in this thread,
 that's a separate issue. I just want to know what you think about
 closing 1.4 support now. If you want to discuss 1.8 quality, start a
 new thread. Thanks.

I don't think it's a separate issue at all.  I would like to see discussion of 
exactly which issues are preventing users from using Asterisk 1.8.  We're 
trying to shift focus to those issues and get them resolved as quickly and as 
efficiently as we can so that we can all move forward.

Resources are limited.  What is the best use of our time to help ensure the 
best future?  Where do we want to see the project in the next 6 months to a 
year?  A primary focus on further solidifying Asterisk 1.8 is what gets us 
there in my mind.

Asterisk 1.4 was released 4.5 years ago.  It mostly just works, and I fully 
expect many to keep using it until they see a need to migrate.  This process 
has been likened to when the community moved from Asterisk 1.2 to 1.4.  
Asterisk 1.8 has been much more stable out of the gate than 1.4, due to many 
things we have done over the years to increase quality, including:

1) We have adopted peer code reviews as common practice for all non-trivial 
changes going into Asterisk.  This alone has _greatly_ increased the quality of 
the code going in.  It is rare that a patch goes up for review where someone 
doesn't point out some sort of problem.  These problems are found and fixed 
_much_ faster in the up front review process than if it had been many months 
later when someone encountered it as a bug in the field.

2) We have placed an increased emphasis on automated testing efforts.  In 
addition to building up a lot of test environments inside of Digium, there is 
now an open source automated testing effort for Asterisk.  There are over 200 
test cases that run every time anyone touches the code.  This includes complex 
call scenarios such as transfers and call parking.  These open source test 
cases touch about 25% of the code (and what it does touch are things we 
considered some of the most important parts).  That is a huge step forward from 
where we started.  We are continuing to place more and more resources on this 
effort to move it forward.

Despite comments in this thread, there _are_ many people using Asterisk 1.8 in 
production, including large installations.  The ones with systems working 
perfectly fine don't tend to make as much noise.  :-)  For those still getting 
hit by problems, I hope that you can make the time to report them so that we 
can work with you to get them resolved.

I started my work on Asterisk as a volunteer 7 years ago and even though it is 
now my full time job, I still put many personal hours into the project.  I care 
very deeply about the success of Asterisk.  I truly believe that the steps we 
have taken with release management are in the best interest of the project.

Thanks,

-- 
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Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Chris Owen

On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote:

 I don't think it's a separate issue at all.  I would like to see discussion 
 of exactly which issues are preventing users from using Asterisk 1.8.  We're 
 trying to shift focus to those issues and get them resolved as quickly and as 
 efficiently as we can so that we can all move forward.

For us the biggest issue is multi-tenant parking not working.   We've really 
given up testing anything beyond that point because without that feature there 
really isn't any way we could use it.

Chris

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Hubris Communications Inc  www.hubris.net
-



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Re: [asterisk-users] [RESOLVED] Echocancellation OSLEC vs MG2 ?


I was right i grab kernel 2.6.35 and build oslec against it and everything 
works!!

Thanks all of you...

-S


From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 28 Apr 2011 14:42:00 +
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?








AH! i dig into kernel and i found there was already echo.ko module exist in 
kernel  

 ls -l /lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko
-rw-r--r-- 1 root root 10840 2011-03-01 21:48 
/lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko

I remove that echo.ko module and install dahdi echo.ko module and it allowed me 
to load oslec module 

I believe i should download kernel-2.6.32.30 and build oslec again that source 
right ? 

-S


From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 28 Apr 2011 14:35:25 +
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?








Thanks for rely,

Actually i have build OSLEC with the help of 
http://www.rowetel.com/blog/?page_id=454 

And now i am getting error at loading module 

root@shirley:~# lsmod | grep echo
dahdi_echocan_mg2   5662  23
dahdi 210313  50 dahdi_echocan_mg2,wanpipe
echo5253  0
root@shirley:~# modprobe dahdi_echocan_oslec
FATAL: Error inserting dahdi_echocan_oslec 
(/lib/modules/2.6.32-30-preempt/dahdi/dahdi_echocan_oslec.ko): Unknown symbol 
in module, or unknown parameter (see dmesg)


What do you suggest ?

-S

Date: Thu, 28 Apr 2011 08:54:30 +0100
From: tiago.ge...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?

linux-dahdi/README has a section on how to compile and install oslec

On 27 April 2011 22:15, Anthony Messina amess...@messinet.com wrote:

On 04/27/2011 02:06 PM, satish patel wrote:

 Which echo cancellation is good between OSLEC and MG2. Dahdi by default use 
 MG2 echo cancellation on channel.  If i want to use OSLEC then what should i 
 need to do ? Do i need to recompile dahdi with OSLEC ?



Yes, you would need to compile the OSLEC kernel module.  Or, if you are

using a RedHat/Fedora based distro, you're welcome to use the

dahdi-linux and dahdi-linux-kmod RPMS I build here.  I include OSLEC

with the dahdi-linux-kmod build.



http://messinet.com/rpms/



--

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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E




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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote:
 
  I don't think it's a separate issue at all. I would like to see
  discussion of exactly which issues are preventing users from using
  Asterisk 1.8. We're trying to shift focus to those issues and get
  them resolved as quickly and as efficiently as we can so that we can
  all move forward.
 
 For us the biggest issue is multi-tenant parking not working. We've
 really given up testing anything beyond that point because without
 that feature there really isn't any way we could use it.

Broken as compared to 1.6.2?  I ask since that feature wasn't in 1.4.

Can you point to a bug report?  I'd like to understand better what's not 
working.

-- 
Russell Bryant
Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org

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[asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Bruce B

Hi everyone,

How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that requires quite some scripting work.

Is there any easy way to simulate a distorted SIP line temporarily for
testing?

I am appreciate experienced inputs.

Thanks
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Chris Owen

On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote:

 For us the biggest issue is multi-tenant parking not working. We've
 really given up testing anything beyond that point because without
 that feature there really isn't any way we could use it.
 
 Broken as compared to 1.6.2?  I ask since that feature wasn't in 1.4.

As compared to 1.6.1.x.   We were using it precisely because we had to have 
multi-tenant parking.

 Can you point to a bug report?  I'd like to understand better what's not 
 working.

https://issues.asterisk.org/view.php?id=18553

Basically for several versions of 1.6.2.x and all 1.8.x that we've tested, when 
you park a call it gets parked in the first parking lot regardless of what 
context the call is in when it is parked.

Chris

--
-
Chris Owen - Garden City (620) 275-1900 -  Lottery (noun):
President  - Wichita (316) 858-3000 -A stupidity tax
Hubris Communications Inc  www.hubris.net
-



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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread David


Hey,

Not sure why you would want to do this. I find nothing destroys a clean 
link like running torrents. Try downloading the 10 most popular torrents 
off of thepiratebay.org ( that are more than 4 gigs ).


( just make sure aren't breaking any copyright rules ).

That'll saturate your link and will give you lots of distortions on your 
VoIP.


David


On 2011-04-28 11:25, Bruce B wrote:

Hi everyone,

How can I introduce some distortion, echo, chopping sound and all 
other bad quality things that can happen to a SIP trunk? I have plenty 
of bandwidth and crisp clear lines so the only thing that I can think 
of is to limit bandwidth but even that requires quite some scripting 
work.


Is there any easy way to simulate a distorted SIP line temporarily for 
testing?


I am appreciate experienced inputs.

Thanks


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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Danny Nicholas

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Sent: Thursday, April 28, 2011 10:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to create distortion, echo, and chopping
sound in a SIP trunk?

Hey,

Not sure why you would want to do this. I find nothing destroys a clean link
like running torrents. Try downloading the 10 most popular torrents off of
thepiratebay.org ( that are more than 4 gigs ).

( just make sure aren't breaking any copyright rules ).

That'll saturate your link and will give you lots of distortions on your
VoIP.

David

On 2011-04-28 11:25, Bruce B wrote: 

Hi everyone, 

How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that requires quite some scripting work. 

Is there any easy way to simulate a distorted SIP line temporarily for
testing?

I am appreciate experienced inputs.

Thanks

[Danny Nicholas] 
Run 4-5 large wgets at once and this will probably do the trick as well -
unless your router has voip prioritized.
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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Stefan Gofferje

On Thursday 28 April 2011, Bruce B wrote:
 
 How can I introduce some distortion, echo, chopping sound and all other bad
 quality things that can happen to a SIP trunk? I have plenty of bandwidth
 and crisp clear lines so the only thing that I can think of is to limit
 bandwidth but even that requires quite some scripting work.
 
 Is there any easy way to simulate a distorted SIP line temporarily for
 testing?
 
 I am appreciate experienced inputs.

Force the switch port which the asterisk is connected to 10MBit/s half-duplex 
and then fire a ping -f -s 65507 asterisk-host from a machine with a 
gigabit-link to the switch.
That should get the line quality pretty much to the bottom.

-S

-- 
 (o_   Stefan Gofferje| SCLT, MCP, CCSA
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler  Koch - the original point and click interface 


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote:
 
  For us the biggest issue is multi-tenant parking not working. We've
  really given up testing anything beyond that point because without
  that feature there really isn't any way we could use it.
 
  Broken as compared to 1.6.2? I ask since that feature wasn't in 1.4.
 
 As compared to 1.6.1.x. We were using it precisely because we had to
 have multi-tenant parking.
 
  Can you point to a bug report? I'd like to understand better what's
  not working.
 
 https://issues.asterisk.org/view.php?id=18553
 
 Basically for several versions of 1.6.2.x and all 1.8.x that we've
 tested, when you park a call it gets parked in the first parking lot
 regardless of what context the call is in when it is parked.

Thanks!  I will take a look at this one and see what we can do.

-- 
Russell Bryant
Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org

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Re: [asterisk-users] how to know status of asterisk from php

2011-04-28 Thread Steve Edwards


Un-top-posting...


On 28 April 2011 07:19, virendra bhati virbh...@gmail.com wrote:


As per you suggestion I write small php scripts but didn't get result. 
Below is the php script and output of programs too.



$priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri);


Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) 
priline=Unable to connect to remote asterisk (does 
/var/run/asterisk.ctl exist?)


On Thu, 28 Apr 2011, Tiago Geada wrote:

The error is pretty straight forward. It is telling you that no Asterisk 
service is running in that machine


Or...

The user that is executing the PHP script does not have write access to 
/var/run/asterisk.ctl.


Is the user executing the PHP script the same user executing the Asterisk 
process?


Changing the permission on /var/run/asterisk.ctl or executing 'sudo 
/usr/sbin/asterisk -rnx pri show spans' may help.


Personally, I think the 'system()' approach is a 'quick and dirty hack' 
and you really should be using AMI.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread David



It also depends on what you want. Torrent will saturate your link, wget 
can do that too, but Torrents always manage to bypass most QoS rules I 
have found on the net, wget doesn't.


So I find torrents always give me a better result for testing my link.

On 2011-04-28 11:36, Danny Nicholas wrote:



*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David

*Sent:* Thursday, April 28, 2011 10:32 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] How to create distortion, echo, and 
chopping sound in a SIP trunk?


Hey,

Not sure why you would want to do this. I find nothing destroys a 
clean link like running torrents. Try downloading the 10 most popular 
torrents off of thepiratebay.org ( that are more than 4 gigs ).


( just make sure aren't breaking any copyright rules ).

That'll saturate your link and will give you lots of distortions on 
your VoIP.


David


On 2011-04-28 11:25, Bruce B wrote:

Hi everyone,

How can I introduce some distortion, echo, chopping sound and all 
other bad quality things that can happen to a SIP trunk? I have plenty 
of bandwidth and crisp clear lines so the only thing that I can think 
of is to limit bandwidth but even that requires quite some scripting 
work.


Is there any easy way to simulate a distorted SIP line temporarily for 
testing?


I am appreciate experienced inputs.

Thanks

  

  
*/[Danny Nicholas]/*

*/Run 4-5 large wgets at once and this will probably do the trick as well -- 
unless your router has voip prioritized./*


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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Tony Mountifield

In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com,
Bruce B bruceb...@gmail.com wrote:
 
 How can I introduce some distortion, echo, chopping sound and all other bad
 quality things that can happen to a SIP trunk? I have plenty of bandwidth
 and crisp clear lines so the only thing that I can think of is to limit
 bandwidth but even that requires quite some scripting work.
 
 Is there any easy way to simulate a distorted SIP line temporarily for
 testing?
 
 I am appreciate experienced inputs.

You could use iptables to cause random packet loss.

See 
http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/
for examples. You might want to precede those rules with ACCEPT rules
for the traffic you want to remain reliable (such as TCP connections).

Cheers
Tony
-- 
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

You can use tc (traffic control) on linux and limit your bandwidth

http://www.linuxtoday.com/infrastructure/2008092400820OSDBNT 

 To: asterisk-users@lists.digium.com
 From: t...@softins.co.uk
 Date: Thu, 28 Apr 2011 15:57:26 +
 Subject: Re: [asterisk-users] How to create distortion, echo, and chopping 
 sound in a SIP trunk?

 In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com,
 Bruce B bruceb...@gmail.com wrote:

  How can I introduce some distortion, echo, chopping sound and all other bad
  quality things that can happen to a SIP trunk? I have plenty of bandwidth
  and crisp clear lines so the only thing that I can think of is to limit
  bandwidth but even that requires quite some scripting work.

  Is there any easy way to simulate a distorted SIP line temporarily for
  testing?

  I am appreciate experienced inputs.

 You could use iptables to cause random packet loss.

 See 
 http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/
 for examples. You might want to precede those rules with ACCEPT rules
 for the traffic you want to remain reliable (such as TCP connections).

 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Administrator TOOTAI


Le 28/04/2011 16:53, Russell Bryant a écrit :

- Original Message -

PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread. Thanks.

I don't think it's a separate issue at all.  I would like to see discussion of 
exactly which issues are preventing users from using Asterisk 1.8.  We're 
trying to shift focus to those issues and get them resolved as quickly and as 
efficiently as we can so that we can all move forward.


Let's see it from another angle: we today are mainly using 1.4 and 1.6.2 
In the last month with faced those regressions, first still not solved:


https://issues.asterisk.org/view.php?id=18539
https://issues.asterisk.org/view.php?id=18998

Do you think we're ready to switch to 1.8 if 1.4/1.6 still have such 
behavior?


As I told in previous answer, we started 1.4 in production very early 
and had lots of troubles, we don't want to face the same over activity 
with 1.8



Resources are limited.

This I understand

What is the best use of our time to help ensure the best future?  Where do we 
want to see the project in the next 6 months to a year?  A primary focus on 
further solidifying Asterisk 1.8 is what gets us there in my mind.


Agree

Asterisk 1.4 was released 4.5 years ago.  It mostly just works, and I fully 
expect many to keep using it until they see a need to migrate.  This process has been 
likened to when the community moved from Asterisk 1.2 to 1.4.  Asterisk 1.8 has been much 
more stable out of the gate than 1.4, due to many things we have done over the years to 
increase quality, including:

[...]

Ok, so why not stay with asterisk 1.4 security *and* bug/regression 
fixes for few weeks/monthes till 1.8 reaches the level that the 
community accept to switch to 1.8


--
Daniel

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[asterisk-users] odbc error - server is gone

2011-04-28 Thread Rizwan Hisham

Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and
here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage.
I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
driver for mysql on the server. I successfully completed the conversion of a
lot of voicemail users into db yesterday. But today on the CLI thsi error
was showing;

[Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL
Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
[Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL
Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
[Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
Execute error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/1757XXX/INBOX']

I know that the error is caused due to stale odbc connection with mysql. But
i want to find out if there is a cure for it. Why the connection went stale
in the first place also.

Any ideas?

-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] how to know status of asterisk from php

2011-04-28 Thread Juan David Diaz

Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl
exist?)

I think, you should start asterisk before executing asterisk commands

regards

Juan.
Linux User #441131


On Thu, Apr 28, 2011 at 1:19 AM, virendra bhati virbh...@gmail.com wrote:

 Hi,

 As per you suggestion I write small php scripts but didn't get result.
 Below is the php script and output of programs too.

 *PHP Script:-*

 ?php
 $priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri);
 $asterisk = system(/etc/init.d/asterisk status, $asterisks);
 $mysql = system(/etc/init.d/mysql status,$mysqls);
 echo priline=.$priline;
 echo br;
 echo pri=.$pri;
 echo br;
 echo asterisk=.$asterisk;
 echo br;
 echo asterisks=.$asterisks;
 echo br;
 echo mysql=.$mysql;
 echo br;
 echo mysqls=.$mysqls;
 echo br;
 ?

 *Output:-*

 Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
 priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl
 exist?)
 pri=1
 asterisk=
 asterisks=127
 mysql=
 mysqls=127



 On Wed, Apr 27, 2011 at 8:43 PM, Juan David Diaz juanch...@gmail.comwrote:

 Hi:

 http://php.net/manual/en/function.system.php

 Then, the commands you shoul run:

 /usr/sbin/asterisk -rnxpri show spans
 /etc/init.d/asterisk status
 /etc/init.d/mysql status
 .
 .
 .
 .
 and so on!!

 good luck!

 Regards.

 Juan.
 Linux User #441131


 On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati virbh...@gmail.comwrote:

 Hi

 How to know status of Asterisk,Mysql. PRI lines and other services from
 PHP scripts ?

 
 Thanks and regards

  Virendra Bhati
 +91-9172341457


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 -

 Thanks and regards

  Virendra Bhati
 +91-9172341457


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


At 05:56 AM 4/28/2011, you wrote:

If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish.  BTW, where does 1.10 fit into this
conversation?


Personally, 1.8 has never lasted more than 12 hours on my box without 
dying and once I figured out how it dies, every beta and every 
release will fail within moments if I followed the same very short 
test script. I did put up a bug report on the problem once and was 
told within moments it wasn't a bug, but I'm not smart enough to 
understand what I'm supposed to do to troubleshoot and the same 
configuration has always run on 1.2, 1.6 and 1.10 so from my 
perspective, it's a bug.


1.10 or trunk as I guess it's currently known has been running on my 
production box for 2 weeks with not one hiccup.


Ira 



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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

Where did you download asterisk 1.10 or trunk ? I search and found nothing. 
could your point me there?

-S

 Date: Thu, 28 Apr 2011 10:06:18 -0700
 To: asterisk-users@lists.digium.com
 From: i...@extrasensory.com
 Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

 At 05:56 AM 4/28/2011, you wrote:
 If I can install 1.8 and
 know that I can turn off things to get to 1.4 solidness, then I don't
 have a problem with this kettle of fish.  BTW, where does 1.10 fit into this
 conversation?

 Personally, 1.8 has never lasted more than 12 hours on my box without 
 dying and once I figured out how it dies, every beta and every 
 release will fail within moments if I followed the same very short 
 test script. I did put up a bug report on the problem once and was 
 told within moments it wasn't a bug, but I'm not smart enough to 
 understand what I'm supposed to do to troubleshoot and the same 
 configuration has always run on 1.2, 1.6 and 1.10 so from my 
 perspective, it's a bug.

 1.10 or trunk as I guess it's currently known has been running on my 
 production box for 2 weeks with not one hiccup.

 Ira 

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Andrew Latham

On Thu, Apr 28, 2011 at 1:34 PM, satish patel satish...@hotmail.com wrote:
 Where did you download asterisk 1.10 or trunk ? I search and found nothing.
 could your point me there?

 -S

svn co http://svn.asterisk.org/svn/asterisk/trunk /usr/src/asterisk_trunk


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Paul Belanger


On 11-04-28 01:06 PM, Ira wrote:

At 05:56 AM 4/28/2011, you wrote:

If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?


Personally, 1.8 has never lasted more than 12 hours on my box without
dying and once I figured out how it dies, every beta and every release
will fail within moments if I followed the same very short test script.
I did put up a bug report on the problem once and was told within
moments it wasn't a bug, but I'm not smart enough to understand what I'm
supposed to do to troubleshoot and the same configuration has always run
on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.

What is the issue number.  Additionally, if you can reproduce this with 
a simple test script, I recommend creating a test for the testsuite and 
posting in on reviewboard.  I'll even talk the time to triage and merge 
the test.


--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


Hey Paul,

We have migrate asterisk from 1.2 to 1.8 in production and we have this issue i 
wouldn't say its critical but just thought point you out. This is open since 
last long time and no one respond :( 

https://issues.asterisk.org/view.php?id=18514

Now i am trying 1.10 and let see whether its going to fix or not. 

-Satish 


 Date: Thu, 28 Apr 2011 13:43:15 -0400
 From: pabelan...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
 
 On 11-04-28 01:06 PM, Ira wrote:
  At 05:56 AM 4/28/2011, you wrote:
  If I can install 1.8 and
  know that I can turn off things to get to 1.4 solidness, then I don't
  have a problem with this kettle of fish. BTW, where does 1.10 fit into
  this
  conversation?
 
  Personally, 1.8 has never lasted more than 12 hours on my box without
  dying and once I figured out how it dies, every beta and every release
  will fail within moments if I followed the same very short test script.
  I did put up a bug report on the problem once and was told within
  moments it wasn't a bug, but I'm not smart enough to understand what I'm
  supposed to do to troubleshoot and the same configuration has always run
  on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.
 
 What is the issue number.  Additionally, if you can reproduce this with 
 a simple test script, I recommend creating a test for the testsuite and 
 posting in on reviewboard.  I'll even talk the time to triage and merge 
 the test.
 
 -- 
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 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org
 
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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Bruce B

Thanks for the input guys. What Tony and Satish suggested are alone the
lines of what I need. It gives me a controlled solution. So, I can change
the level of distortion as I please. Using tc I pretty much killed the line
to the point I wasn't able to receive call and terminal was really slow as
well. I am going to try the the packet drop method now. I think that is the
right one for the situation.

Thanks again

On Thu, Apr 28, 2011 at 11:57 AM, Tony Mountifield t...@softins.co.ukwrote:

In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com,
Bruce B bruceb...@gmail.com wrote:

How can I introduce some distortion, echo, chopping sound and all other
bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that requires quite some scripting work.

Is there any easy way to simulate a distorted SIP line temporarily for
testing?

I am appreciate experienced inputs.

You could use iptables to cause random packet loss.

See
http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/
for examples. You might want to precede those rules with ACCEPT rules
for the traffic you want to remain reliable (such as TCP connections).

Cheers
Tony
--
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Re: [asterisk-users] odbc error - server is gone

2011-04-28 Thread Sherwood McGowan

On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 Hi list,
 yesterday I converted my voicemail.conf to realtime voicemail and also
 configured to store the voicemessages in a database using odbc as described
 here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and
 here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage.
 I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
 driver for mysql on the server. I successfully completed the conversion of a
 lot of voicemail users into db yesterday. But today on the CLI thsi error
 was showing;

 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
 SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
 SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
 Execute error!
 [SELECT COUNT(*) FROM voicemessages WHERE dir =
 '/var/spool/asterisk/voicemail/default/1757XXX/INBOX']

 I know that the error is caused due to stale odbc connection with mysql.
 But i want to find out if there is a cure for it. Why the connection went
 stale in the first place also.

 Any ideas?

 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.

 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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do you have sanitysql = select 1 configured in res_odbc.ini?

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Re: [asterisk-users] Best modem for chan_datacard

2011-04-28 Thread Tiago Geada

I used succesfully huawei E1550

On 24 April 2011 16:45, Dovid Bender asteriskus...@dovid.net wrote:

  Hi List,

 I am looking to play around with chan_datacard. Any advice on the best
 device to test with (that I can find on eBay) ?

 Regards,

 Dovid


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
 Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
 few weeks/monthes till 1.8 reaches the level that the community accept to 
 switch
 to 1.8

What is the guide here? What is the level that the community accepts?
Unfortunately that is a statement that is impossible to measure quantitatively.
The answer will always be, We're not ready!

Having to focus on issues on both the 1.4 and 1.8 branches simultaneously
distracts from the goal of making 1.8 stable (which in my several deployments
recently, it seems to be).

I've also seen very few issues being committed to 1.4 for quite some time, which
seems to tell me 1.4 is stable for most deployments. It's not like there has
been a flurry of activity around 1.4 and all of a sudden it's being cut off. In
my estimation the number of commits to 1.4 going from a few to none is not a
significant direction change.

Asterisk 1.4 isn't going away. The code base won't stop working on your system
-- it will continue happily plugging away as it always has. The code will
continue to be available for deployments.

With focus being directed to 1.8, the issues that may be blocking you from
having a successful migration to, or deployment of, Asterisk 1.8 will get fixed
that much sooner.

If the community won't, or can't, step up to maintain a community based branch
which has very few changes being made to it, then I'm not sure it is fair to
expect Digium to do that.

Leif.


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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

2011-04-28 Thread Andreas Sikkema

On 4/28/11 5:25 PM, Bruce B wrote:
 Is there any easy way to simulate a distorted SIP line temporarily for
 testing?

Build a Linux based router and use netem/tc to mess around with the
routed traffic. You can insert packetloss, jitter, etc and have it be
reproducable.

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[asterisk-users] anybody out there sucessfully using gnugk?

2011-04-28 Thread Danny Nicholas

Hi List,
I have a client that wants me to replace their existing H323
gateway.  I am able to get ooh323 and h323 to work fine in a native
environment, but the whole thing goes to heck when I have to cross networks.
Gnugk seems to be the answer to this, but I can't seem to get it to work
right.  Any ideas?

Thanks
Danny Nicholas


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Administrator TOOTAI


Le 28/04/2011 21:47, Leif Madsen a écrit :

On 11-04-28 12:04 PM, Administrator TOOTAI wrote:

Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to switch
to 1.8

What is the guide here? What is the level that the community accepts?
Unfortunately that is a statement that is impossible to measure quantitatively.
The answer will always be, We're not ready!


Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: 
till 1.8 is not at the feature level and stability of 1.4, people like 
me will not move to 1.8 Measure is easy :-)



Having to focus on issues on both the 1.4 and 1.8 branches simultaneously
distracts from the goal of making 1.8 stable (which in my several deployments
recently, it seems to be).


Again, I think that maintaining 1.4 on his today level is ok *if and 
only if* bugs/regression are taking in account, not only security.



[...]

With focus being directed to 1.8, the issues that may be blocking you from
having a successful migration to, or deployment of, Asterisk 1.8 will get fixed
that much sooner.


In production you can't use something which will be fixed sooner. It 
has to work straight on, at least when you upgrade from a previous 
version. Customer doesn't care if the new version is more up to date and 
has new features if in the mean time they don't have features that they 
had before.



If the community won't, or can't, step up to maintain a community based branch
which has very few changes being made to it, then I'm not sure it is fair to
expect Digium to do that.


That's one point for you: community seems to say we want that 1.4 still 
lives but no one [want|doesn't have the knowledge] to participate on 
maintaining the community branch.


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


At 10:43 AM 4/28/2011, you wrote:

On 11-04-28 01:06 PM, Ira wrote:

At 05:56 AM 4/28/2011, you wrote:

If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?


Personally, 1.8 has never lasted more than 12 hours on my box without
dying and once I figured out how it dies, every beta and every release
will fail within moments if I followed the same very short test script.
I did put up a bug report on the problem once and was told within
moments it wasn't a bug, but I'm not smart enough to understand what I'm
supposed to do to troubleshoot and the same configuration has always run
on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.
What is the issue number.  Additionally, if you can reproduce this 
with a simple test script, I recommend creating a test for the 
testsuite and posting in on reviewboard.  I'll even talk the time to 
triage and merge the test.



The test is this:

Pick up one of my 3 Aastra sip phones.
Dial 11 to get dial tone on a POTS line connected to a TDM04
Dial the POTS line connected to port 2 on that same TDM04
Call goes directly to voice mail and I get a bunch of SIP 
re-transmission errors.


Then I get told to read the SIP retransmission document which might 
as well be written in Greek for all the good it does me.
I've no clue what the bug report number is, it was back around RC1 or 
2.  If you search the archives for my email you will probably find my 
posts about this. I don't post much so they should stand out.


If you want to look at this with my help, an email off-list will get 
your use of me and my Asterisk box.


Ira


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Re: [asterisk-users] anybody out there sucessfully using gnugk?

2011-04-28 Thread Andreas Sikkema

On 4/28/11 10:30 PM, Danny Nicholas wrote:
 Hi List,
 I have a client that wants me to replace their existing H323
 gateway.  I am able to get ooh323 and h323 to work fine in a native
 environment, but the whole thing goes to heck when I have to cross networks.
 Gnugk seems to be the answer to this, but I can't seem to get it to work
 right.  Any ideas?

It's been years since I used GNUGk, but I'd check the mailinglist at
http://www.gnugk.org/ The core developers have always been very helpful
to me.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-04-28 04:33 PM, Administrator TOOTAI wrote:
 Le 28/04/2011 21:47, Leif Madsen a écrit :
 On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
 Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes 
 for
 few weeks/monthes till 1.8 reaches the level that the community accept to 
 switch
 to 1.8
 What is the guide here? What is the level that the community accepts?
 Unfortunately that is a statement that is impossible to measure 
 quantitatively.
 The answer will always be, We're not ready!
 
 Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: till
 1.8 is not at the feature level and stability of 1.4, people like me will not
 move to 1.8 Measure is easy :-)

But that's what I don't get. No one is *forcing* you to move to 1.8 *right now*.
The code base for 1.4 isn't going anywhere. Anyone is able to keep deploying 1.4
(or 1.2, or 1.0, or 0.9 for that matter) to their hearts content.

Leif.

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Re: [asterisk-users] how to know status of asterisk from php


On 28/04/11 10:53 PM, virendra bhati wrote:

Hi,

I am running this script from wabsite and want to make it just like
FreePBX show status information of  Asterisk, mysql etc.

*Asterisk is running as root* at my end.

When I start script from command prompt then I am getting error message..

[root@cent68 mtnl]# php temp.php
PHP Parse error:  syntax error, unexpected ',' in
/var/www/html/mtnl/temp.php on line 3

Parse error: syntax error, unexpected ',' in /var/www/html/mtnl/temp.php
on line 3


You have an error on line 3 of your code - probably somewhere around the 
comma.


--
Cheers,

Matt Riddell
___

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http://www.venturevoip.com/cc.php (Call Centre Solutions)

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Administrator TOOTAI


Le 28/04/2011 22:43, Leif Madsen a écrit :

On 11-04-28 04:33 PM, Administrator TOOTAI wrote:

Le 28/04/2011 21:47, Leif Madsen a écrit :

On 11-04-28 12:04 PM, Administrator TOOTAI wrote:

Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to switch
to 1.8

What is the guide here? What is the level that the community accepts?
Unfortunately that is a statement that is impossible to measure quantitatively.
The answer will always be, We're not ready!

Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: till
1.8 is not at the feature level and stability of 1.4, people like me will not
move to 1.8 Measure is easy :-)

But that's what I don't get. No one is *forcing* you to move to 1.8 *right now*.
The code base for 1.4 isn't going anywhere. Anyone is able to keep deploying 1.4
(or 1.2, or 1.0, or 0.9 for that matter) to their hearts content.


Sure. Please follow the 2 next stories:

- had a customer running 1.4.26 We upgraded to a new server and 
installed 1.4.39, last version at this time. Bang: voicemail doesn't 
work as it should, had to fallback to 1.4.26 Customer is still running 
this version.
- have 1.4.41 and 1.6.16 which are no more able to use auth keys in iax 
since we update one server from 1.4 to 1.6


Now imagine that 1.4 stays at only security level. For first case we 
have 2 options: upgrading for security reasons to last version but then 
no more voicemail, or staying with 1.4.26. In the second case, upgrading 
both servers to test with 1.8. If it's still not working, it was time 
loose beside other problems.


Yes, we have servers for testing, but really, who would think that such 
2 problems araised with an 1.4 stable version? Same was few versions 
before (1.4.20~1.4.28 if I good remember) with attempted call transfer: 
was working on one version, stop to next one, worked again aso. Even in 
a test environment you can't simulate all setups.


Hope that this both scenario gives you a new vision ;-) and why I tell 
that bugs and regressions should be taken in account at the same level 
as security.


--
Daniel

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Paul Belanger


On 11-04-28 04:35 PM, Ira wrote:

If you want to look at this with my help, an email off-list will get
your use of me and my Asterisk box.

I just posted a patch on the issue tracker, I'll need to get it reviewed 
to see if this is the best approach.


--
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


At 03:22 PM 4/28/2011, you wrote:

On 11-04-28 04:35 PM, Ira wrote:

If you want to look at this with my help, an email off-list will get
your use of me and my Asterisk box.
I just posted a patch on the issue tracker, I'll need to get it 
reviewed to see if this is the best approach.



I would comment that I've been complaining about this since RC1 or 2 
and if you just fixed it in 2 hours that there is something seriously 
wrong with the bug tracking system.  I mean, I reported it a long 
time ago and while it was probably not the best bug report ever, I 
would have been more than willing to do almost anything to help fix 
it. I know what beta tester means, I've beta tested disk defraggers 
and disk caches and lost everything when they had the wrong bug and I 
know it can take a few tries to both fix the bug and for someone to 
help me identify it so they have an idea of where to look. Personally 
I'd just assumed that 1.8 was going to stay broken as no one seemed 
to care and was really happy when trunk worked as that meant I could 
move on. I like the bleeding edge and will always run the current 
beta on my small system unless I find a problem. I know it's 
dangerous, but it gives me the best chance of influencing where the 
product is going. Not much chance with this, but old habits die hard.


And thanks for looking at this.  The offer to help stands. Once UPS 
picks up in the afternoon, 3:30PM PST, I'm happy to try anything for you.


Ira


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 Sure. Please follow the 2 next stories:
 
 - had a customer running 1.4.26 We upgraded to a new server and
 installed 1.4.39, last version at this time. Bang: voicemail doesn't
 work as it should, had to fallback to 1.4.26 Customer is still running
 this version.
 - have 1.4.41 and 1.6.16 which are no more able to use auth keys in
 iax
 since we update one server from 1.4 to 1.6
 
 Now imagine that 1.4 stays at only security level. For first case we
 have 2 options: upgrading for security reasons to last version but
 then no more voicemail, or staying with 1.4.26. In the second case,
 upgrading both servers to test with 1.8. If it's still not working, it was 
 time
 loose beside other problems.

If there are obvious regressions in major functionality such as voicemail, I'm 
more than happy to still consider making fixes for those problems during the 
security maintenance period.  It has to be pretty clear, though, and in this 
particular case, it is.

Can you point to the bug number please?  I want to make sure this voicemail 
problem is resolved as soon as possible.

-- 
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Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 Thanks Matt.
 
 There seems to be an unresolved deadlock since the birth of 1.8.
 Using the most basic feature of a PBX, try to pickup some elses
 ringing
 extension - DEADLOCK.
 
 But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and
 it's
 more uptodate review https://reviewboard.asterisk.org/r/1185/

Thanks, Alec.  I have added this to the roadmap for the next 1.8 update.  I'll 
make sure it gets resolved before then.

-- 
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


On 29/04/11 10:10 AM, Alec Davis wrote:

Thanks Matt.

There seems to be an unresolved deadlock since the birth of 1.8.
Using the most basic feature of a PBX, try to pickup some elses ringing
extension - DEADLOCK.

But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and it's
more uptodate review https://reviewboard.asterisk.org/r/1185/


Yeah, not sure why that one's not affecting me.

I'm using the Set(_PICKUPMARK=1) thingy with a little bit of logic from 
DB functions and customers don't seem to be hitting it.


I'm not using the *8 thing though.

Yeah, just checked one system and they're definitely using it:

/var/log/asterisk/cdr-custom# grep Pickup Master.csv |wc -L
196

196 times since I upgrade them on the 11th of February.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


- Original Message -
 I would comment that I've been complaining about this since RC1 or 2
 and if you just fixed it in 2 hours that there is something seriously
 wrong with the bug tracking system. I mean, I reported it a long
 time ago and while it was probably not the best bug report ever, I
 would have been more than willing to do almost anything to help fix
 it. I know what beta tester means, I've beta tested disk defraggers
 and disk caches and lost everything when they had the wrong bug and I
 know it can take a few tries to both fix the bug and for someone to
 help me identify it so they have an idea of where to look. Personally
 I'd just assumed that 1.8 was going to stay broken as no one seemed
 to care and was really happy when trunk worked as that meant I could
 move on. I like the bleeding edge and will always run the current
 beta on my small system unless I find a problem. I know it's
 dangerous, but it gives me the best chance of influencing where the
 product is going. Not much chance with this, but old habits die hard.

I don't think there's anything inherently wrong with the bug tracking system.  
It's more of a resource issue with many conflicting priorities.  Officially 
letting off some of the pressure from older branches does help.  I would like 
to be making faster progress through bug reports and patches.  I do have an 
open position for another full time Asterisk developer at Digium in case anyone 
is interested.  :-)

-- 
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Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


At 03:48 PM 4/28/2011, you wrote:


- Original Message -
 I would comment that I've been complaining about this since RC1 or 2
 and if you just fixed it in 2 hours that there is something seriously
 wrong with the bug tracking system. I mean, I reported it a long
 time ago and while it was probably not the best bug report ever, I
 would have been more than willing to do almost anything to help fix
 it. I know what beta tester means, I've beta tested disk defraggers

I don't think there's anything inherently wrong with the bug 
tracking system.  It's more of a resource issue with many 
conflicting priorities.  Officially letting off some of the pressure 
from older branches does help.  I would like to be making faster 
progress through bug reports and patches.  I do have an open 
position for another full time Asterisk developer at Digium in case 
anyone is interested.  :-)


OK, maybe not, but if I thought it was a bug and you discover it was 
a bug and fix it, than who was it who decided it wasn't a bug 15 
minutes after I put it in the bug tracker and why did that person 
have that much power?  Look, I know things take time to fix and test, 
I have no problem with that and I know users report things that 
aren't bugs as bugs.  I develop software and my users do all those 
annoying things too, but I can't slap them down like that if I expect 
them to continue being customers. And I know the people who do this 
are volunteers, but my software is free, so I'm a volunteer too.


Look, I'm not complaining, I'm happy with trunk and I don't care any 
more if 1.8 ever works. If it was up to me I'd say abandon it and 
move on, but it's not up to me. I only brought it up again because of 
the thread about the usability of 1.8.


Asterisk made an amazing change in my life and solved problems in 
ways I never imagined possible before accidently discovering it 5 years ago.


If nothing else, the ability to not have any phone but my wife's ring 
when the annoying members of her family call is worth every penny I 
spent on the hardware.


Ira 



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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Paul Belanger


On 11-04-28 06:39 PM, Ira wrote:

I would comment that I've been complaining about this since RC1 or 2 and
if you just fixed it in 2 hours that there is something seriously wrong
with the bug tracking system.  I mean, I reported it a long time ago and
while it was probably not the best bug report ever, I would have been
more than willing to do almost anything to help fix it.


No, I don't believe the issue tracker is seriously broken but understand 
there are over 920 open issues at the moment. To be honest, I only 
looked at the code because of a personal interest to learn more about 
chan_sip.c and because of the power outages happening at Digium.


jsmith in #asterisk-dev summed it up best a few months ago:

It's open source software -- so if you want a change made, you have 
three basic choices:

   1) Scratch your own itch
   2) Pay someone else to scratch your itch
   3) Convince someone else that it's their itch as well, and be 
patient until they scratch your itch



I know what beta
tester means, I've beta tested disk defraggers and disk caches and lost
everything when they had the wrong bug and I know it can take a few
tries to both fix the bug and for someone to help me identify it so they
have an idea of where to look. Personally I'd just assumed that 1.8 was
going to stay broken as no one seemed to care and was really happy when
trunk worked as that meant I could move on.


+1 for testers.  Sometimes the majority of the work is just reproducing 
the issue. And I said this before, if you can reproduce the issue and 
automate it, I'll take the required steps to help merge the test into 
the testsuite.


--
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Orginate not working well with PSTN lines

2011-04-28 Thread Gilles

On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

I guess it was a lot of work, and nobody bothered adding this to the
Zaptel driver.


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Alec Davis

Making an assumption here, I'm sure I cleared the remaining resequencing
issues up in 1.4 SVN and 1.6.2 SVN.
https://issues.asterisk.org/view.php?id=19032

The issues I uncovered and fixed were when a new voicemail is left, while a
mailbox is open for review and the user deletes a message.

Alec


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
 Russell Bryant
 Sent: Friday, 29 April 2011 10:42 a.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Discussion: Are we ready to 
 leave 1.4 behind?
 
 
 - Original Message -
  Sure. Please follow the 2 next stories:
  
  - had a customer running 1.4.26 We upgraded to a new server and 
  installed 1.4.39, last version at this time. Bang: 
 voicemail doesn't 
  work as it should, had to fallback to 1.4.26 Customer is 
 still running 
  this version.
  - have 1.4.41 and 1.6.16 which are no more able to use auth keys in 
  iax since we update one server from 1.4 to 1.6
  
  Now imagine that 1.4 stays at only security level. For 
 first case we 
  have 2 options: upgrading for security reasons to last version but 
  then no more voicemail, or staying with 1.4.26. In the second case, 
  upgrading both servers to test with 1.8. If it's still not 
 working, it 
  was time loose beside other problems.
 
 If there are obvious regressions in major functionality such 
 as voicemail, I'm more than happy to still consider making 
 fixes for those problems during the security maintenance 
 period.  It has to be pretty clear, though, and in this 
 particular case, it is.
 
 Can you point to the bug number please?  I want to make sure 
 this voicemail problem is resolved as soon as possible.
 
 --
 Russell Bryant
 Digium, Inc.   |   Engineering Manager, Open Source Software
 445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
 www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org
 
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Re: [asterisk-users] asterisk practices

2011-04-28 Thread Jan Bakuwel

Hi Vip,

On 28/04/11 05:34, vip killa wrote:
 I just completed building a feature rich asterisk voicemail system
 using perl, php, and mysql.
 My only concern is that the system i built will not be able to handle
 the call volume needed. Let me start by explaining my setup.

 Incoming call - route.agi (perl - mysql lookup) - AGI -
 voicemailbox (using mysql odbc) or terminate with wrong number message

 if a message is left in a voicemailbox the following happens:
 externnotify - notify.pl http://notify.pl (perl - mysql lookup) -
 up to 2 calls originated (using AMI), up to 4 emails sent out (with up
 to 2 attachemnts of voicemail)

 this system may need to handle up to 50 concurrent calls. the
 notify.pl http://notify.pl script may be called several times a
 second.
 My question is, will asterisk be able to handle calling the notify.pl
 http://notify.pl script that many times? or is there a better way
 to handle large volumes of voicemail notification, thank you in
 advance for your input.

How long is a piece of string? :-)

It all depends on the performance of your hardware.

Generally speaking external scripts do have a performance impact so if
you find out that your system won't handle the amount of concurrent
calls you can either find the bottleneck in your hardware (most likely
disc I'd say) and fix that or move logic from your external scripts to
the dialplan.

cheers,
Jan

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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

2011-04-28 Thread Jan Bakuwel

Hi,

I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian  Ubuntu are available from packages.asterisk.org.
Observing some recent discussions on this list, it seems that 1.8 might
not yet be ready for production use. Would whoever kindly makes the
Asterisk 1.8 packages available also consider doing that for 1.6
releases? If the build environment has been set up for 1.8, I'd imagine
it would be easy to set up something similar for 1.6 releases?

kind regards,
Jan


On 27/04/11 05:04, Asterisk Development Team wrote:
 The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.
 This release is available for immediate download at
 http://downloads.asterisk.org/pub/telephony/asterisk/

 The release of Asterisk 1.6.2.18 resolves several issues reported by the
 community and would have not been possible without your participation.
 Thank you!

 The following is a sample of the issues resolved in this release:

  * Only offer codecs both sides support for directmedia.
(Closes issue #17403. Reported, patched by one47)

  * Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchuan, grecco. Patched by rmudgett)
NOTE: Be sure to read the ChangeLog for more information about these 
 changes.

  * Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler)

  * Fix channel redirect out of MeetMe() and other issues with channel 
 softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
Patched by russellb)

  * Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
jpeeler)

  * Guard against retransmitting BYEs indefinitely during attended transfers 
 with
chan_sip.
(Review: https://reviewboard.asterisk.org/r/1077/)

 In addition to the changes listed above, commits to resolve security issues
 AST-2011-005 and AST-2011-006 have been merged into this release. More
 information about AST-2011-005 and AST-2011-006 can be found at:

 http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
 http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

 For a full list of changes in this release, please see the ChangeLog:

 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18

 Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available


On 29/04/11 11:19 AM, Jan Bakuwel wrote:

Hi,

I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian  Ubuntu are available from packages.asterisk.org.
Observing some recent discussions on this list, it seems that 1.8 might
not yet be ready for production use. Would whoever kindly makes the
Asterisk 1.8 packages available also consider doing that for 1.6
releases? If the build environment has been set up for 1.8, I'd imagine
it would be easy to set up something similar for 1.6 releases?


You shouldn't put *any* system into production unless you have a clear 
list of what features you will be providing, and have a way of testing 
that those features work :-)


If you do this then every update can be tested to work with those 
features, and a customer's system shouldn't crash, no matter what 
version you're using.  As I've said 1.8 is working under these 
circumstances for me in production.


One thing I'll note though is that as time goes on and you get better at 
these types of things you'll come up with some pretty crazy tests - and 
still customers will do things you couldn't possibly have thought to test.


So, long story short I recommend:

1. Make a list of the applications and modules you'll be using and a 
list of ways they'll be used.


2. Disable everything else

3. Test these apps/functions in the most intense way you can think of

4. Move the system to production.

The thing here is that if you're able to provide the same system to 
multiple customers then it doesn't end up being such a crazy list of 
things to check.


--
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Matt Riddell
___

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-04-28 07:02 PM, Ira wrote:
 At 03:48 PM 4/28/2011, you wrote:
 OK, maybe not, but if I thought it was a bug and you discover it was a bug and
 fix it, than who was it who decided it wasn't a bug 15 minutes after I put it 
 in
 the bug tracker and why did that person have that much power?  Look, I know
 things take time to fix and test, I have no problem with that and I know users
 report things that aren't bugs as bugs.  I develop software and my users do 
 all
 those annoying things too, but I can't slap them down like that if I expect 
 them
 to continue being customers. And I know the people who do this are volunteers,
 but my software is free, so I'm a volunteer too.

Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes things
accidentally get closed if it looks like a configuration issue.

If anyone ever opens an issue they they feel is a bug and the issue is closed,
then the best forum is the #asterisk-bugs IRC channel. This allows you to speak
with the bug marshals and to work through some additional information that might
be required to help determine that something is truly an issue.

Leif.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Ernie Dunbar

 On 29/04/11 5:06 AM, Ira wrote:
 At 05:56 AM 4/28/2011, you wrote:
 If I can install 1.8 and
 know that I can turn off things to get to 1.4 solidness, then I
 don't
 have a problem with this kettle of fish. BTW, where does 1.10 fit into
 this
 conversation?

 Personally, 1.8 has never lasted more than 12 hours on my box without
 dying and once I figured out how it dies, every beta and every release
 will fail within moments if I followed the same very short test script.
 I did put up a bug report on the problem once and was told within
 moments it wasn't a bug, but I'm not smart enough to understand what I'm
 supposed to do to troubleshoot and the same configuration has always run
 on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.

 What's the URL to the bug you submitted?

 I'm running 1.8 here 24/7 with no problems other than the ones that Alec
 Davis fixed.  I've got it running in I think 4 installations and we're
 not getting any core dumping or anything - obviously I'm only using a
 subset of the full functionality and most modules are not included.

What features do you have disabled? It would be helpful to know this for
future 1.8 implementation, although right now we can't quite use it yet.


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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

2011-04-28 Thread Jan Bakuwel

Hi Matt,


On 29/04/11 11:26, Matt Riddell wrote:
 On 29/04/11 11:19 AM, Jan Bakuwel wrote:
 Hi,

 I'm about to deliver a production system based on Debian Squeeze and
 Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
 packages for Debian  Ubuntu are available from packages.asterisk.org.
 Observing some recent discussions on this list, it seems that 1.8 might
 not yet be ready for production use. Would whoever kindly makes the
 Asterisk 1.8 packages available also consider doing that for 1.6
 releases? If the build environment has been set up for 1.8, I'd imagine
 it would be easy to set up something similar for 1.6 releases?

 You shouldn't put *any* system into production unless you have a clear
 list of what features you will be providing, and have a way of testing
 that those features work :-)

 If you do this then every update can be tested to work with those
 features, and a customer's system shouldn't crash, no matter what
 version you're using.  As I've said 1.8 is working under these
 circumstances for me in production.

 One thing I'll note though is that as time goes on and you get better
 at these types of things you'll come up with some pretty crazy tests -
 and still customers will do things you couldn't possibly have thought
 to test.

 So, long story short I recommend:

 1. Make a list of the applications and modules you'll be using and a
 list of ways they'll be used.

 2. Disable everything else

 3. Test these apps/functions in the most intense way you can think of

 4. Move the system to production.

 The thing here is that if you're able to provide the same system to
 multiple customers then it doesn't end up being such a crazy list of
 things to check.


All valid points.

As you say users will still do things I couldn't have possibly imagined.
And some bugs, deadlocks conditions in real time systems for instance
might well be particularly hard to find or write test plans for. The
release notes for 1.6.2.18 state that  some issues (such as deadlocks)
found in previous releases have been addressed.

You didn't answer my question though :-)

Rather than testing and finding issues that have already been resolved,
I'd prefer to have an efficient way to upgrade Asterisk to released
versions. A package system provides an efficient way to do this. The
fact that something like packages.asterisk.org exists seems to prove my
point. Upgrading the system obviously doesn't mean you won't have to do
any testing but it should make the testing more efficient - at least for
stable releases.

cheers,
Jan


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-04-28 07:09 PM, Alec Davis wrote:
 Making an assumption here, I'm sure I cleared the remaining resequencing
 issues up in 1.4 SVN and 1.6.2 SVN.
 https://issues.asterisk.org/view.php?id=19032
 
 The issues I uncovered and fixed were when a new voicemail is left, while a
 mailbox is open for review and the user deletes a message.

Can anyone who has this issue currently please test the 1.4 branch? Feedback
would be extremely helpful in determining if anything further needs to be done
here. If so, then please open a new issue and report here.

Leif.


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


On 29/04/11 11:51 AM, Ernie Dunbar wrote:

On 29/04/11 5:06 AM, Ira wrote:

At 05:56 AM 4/28/2011, you wrote:

If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I
don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?


Personally, 1.8 has never lasted more than 12 hours on my box without
dying and once I figured out how it dies, every beta and every release
will fail within moments if I followed the same very short test script.
I did put up a bug report on the problem once and was told within
moments it wasn't a bug, but I'm not smart enough to understand what I'm
supposed to do to troubleshoot and the same configuration has always run
on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.


What's the URL to the bug you submitted?

I'm running 1.8 here 24/7 with no problems other than the ones that Alec
Davis fixed.  I've got it running in I think 4 installations and we're
not getting any core dumping or anything - obviously I'm only using a
subset of the full functionality and most modules are not included.


What features do you have disabled? It would be helpful to know this for
future 1.8 implementation, although right now we can't quite use it yet.


The opposite of what we're using :-)

We've been reworking our GUI software to work on embedded systems as 
well as larger so we use:


AGI for all outbound calling logic - our licensing code sets up routes 
for the customers (i.e. which providers they're using etc) and then they 
chose order (i.e. VoIP followed by Analogue etc).  If a destination 
can't be matched via an outbound route then the call is passed back to 
the dialplan.


Applications/Functions we use:

Macro, Dial, VoiceMail, VoiceMailMain, Goto, GotoIf, GotoIfTime, Hangup, 
UserEvent, Answer, Playback, Record


Some others:

* Pickup application with PICKUPMARK
* DB Functions
* We don't use Asterisk Realtime for these systems
* Call transfers etc are all done by the phones themselves
* DAHDI for timing - even if it's just DAHDI_dummy
* MeetMe (we haven't started using confbridge yet)
* Set application for variables
* hints
* SIPAddHeader or Set(__SIPADDHEADER=
* Outbound calling via IAX2, DAHDI and SIP - depending on the customer
* RFC2833 or Inband DTMF (depending on issues)

And that's it.

We don't use any of the imap voicemail stuff, don't usually use Google 
Talk or anything.  Don't usually use Jabber. Try to stay away from Local 
channels wherever possible.  Restart Asterisk in the middle of the night 
in case there are any memory leaks.


If we ever have any problems we try to track it down to the exact 
revision that caused the problem, read the commit and try and submit a 
bug entry with as much detail as possible.  It's pretty unusual for you 
to be the only person experiencing a bug so normally if you come across 
something you'll see other people with the same problem.  If you don't 
it's because you're doing something different to the majority of users 
or it's a very new bug.  So you first look at what you're doing that's 
different (we use chan_lcr occasionally as BRI isn't working for us with 
DAHDI - LCR has caused some issues).  If it is caused by doing something 
in a way that is different then see if you can do it how most people 
would.  If it still causes an issue, either fix it or submit a ticket. 
You can usually work around most things.


For example we had a problem last week where an incoming call to a DDI 
had a 302 redirect from the phone to another number - i.e. the person 
was out of the office so they redirected to their cell.  When the call 
went back to Asterisk it used the local channel and made an outbound 
call to the cellphone.  After 2 seconds of ringing it would hangup and 
head back to the desk phone - that would redirect it back to the 
cellphone etc etc.


It turned out that for whatever reason the LCR channel wasn't happy with 
the redirection - when tested with the incoming call coming from IAX 
instead of LCR it worked fine.  We then thought that maybe it was 
because the LCR channel hadn't been answered.  We added an Answer() 
before sending the call to the phone and it resolved the problem.


This was not a crash and was caused by the fact that we were doing 
something that most people aren't (using chan_lcr in Asterisk 1.8).  If 
everyone's calls did this when they saw a 302 redirect it certainly 
would have shown up on the issue tracker.


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


At 04:49 PM 4/28/2011, you wrote:

Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes things
accidentally get closed if it looks like a configuration issue.

If anyone ever opens an issue they they feel is a bug and the issue is closed,
then the best forum is the #asterisk-bugs IRC channel. This allows 
you to speak
with the bug marshals and to work through some additional 
information that might

be required to help determine that something is truly an issue.


Well, I've no idea how to do that. I can duplicate the problem every 
time on my system in one step, but I have no idea how to suggest you 
test it or if it requires my particular configuration or Aastra 
phones.  I can do almost anything in Windows, most anything on a Mac 
and almost nothing on a Linux box except make and install Asterisk 
from source and edit the Asterisk configuration files.  You often 
make the assumption that just because I use Asterisk I know Linux. It 
would be nice, but I've made my living doing DOS and then Windows 
support since DOS 2.1 and for me, Linux is just something I need to 
run my Asterisk box.


I'm happy to help, but I need help to do it.

Ira 



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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Alec Davis

Let me try to better describe the test senario that I found, and have been
commited to 1.4svn, 1.6.2svn 1.8svn and trunk.
All aspects need to be thrased out though.

Leave Phone-A 2 new messages, and for this example we only have 2 new
messages.

Now to create the problem - (gaps in the message sequence):

1). From Phone-A, Enter voicemail and start listening to your 1st message.
2). From Phone-B ring Phone-A, which should go to voicemail
Leave a message. Do it again. So now we have new 2 messages, on top of
the initial 2, a total of 4.
3). At Phone-A delete the 1st message, and you now should have 1 left (we
don't know about the 2 new ones). DON'T hangup.
4). Hangup Phone A.
On closing mailbox the resequence only knew about 2 messages, not 4,
thus the message sequence became 0, 2 and 3.
5). DON'T open the mailbox yet, as the openmailbox resequence will fix it
up.

This is where the problems start - (further messages start to get lost
forever), only fixed if the user goes into their mailbox;

6). From Phone-A ring a test extension, Voicemail will do, but don't enter
your BOX number.
7). From Phone-B ring Phone-A, which goes to voicmail, leave a message, do
it again. We now have 2 new message, total should now be 5.
8). Hangup Phone-A

The result:
9). From Phone-A go into voicemail, we should have 5 new messages.
If there are 4, then we have lost some messages.
If there are 5, then it looks well, but still continue testing. This is
just one of many senarios.

Alec Davis

 

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
 Leif Madsen
 Sent: Friday, 29 April 2011 12:03 p.m.
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Discussion: Are we ready to 
 leave 1.4 behind?
 
 On 11-04-28 07:09 PM, Alec Davis wrote:
  Making an assumption here, I'm sure I cleared the remaining 
  resequencing issues up in 1.4 SVN and 1.6.2 SVN.
  https://issues.asterisk.org/view.php?id=19032
  
  The issues I uncovered and fixed were when a new voicemail is left, 
  while a mailbox is open for review and the user deletes a message.
 
 Can anyone who has this issue currently please test the 1.4 
 branch? Feedback would be extremely helpful in determining if 
 anything further needs to be done here. If so, then please 
 open a new issue and report here.
 
 Leif.
 
 
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


On 29/04/11 1:16 PM, Ira wrote:

Well, I've no idea how to do that. I can duplicate the problem every


IRC is an online chat system like MSN or Skype except that it's more 
like a mailing list - you can talk to lots of people at the same time.


On Windows you can use a program like mIRC to connect to 
irc.freenode.net or even a plugin in Firefox.


Once you're connected to IRC you can join chat rooms.

There are some like #asterisk for discussion about Asterisk and 
#asterisk-bugs for discussion about Asterisk bugs.


Post back here if you have any problems connecting.

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[asterisk-users] Friday on VUC: Jabber/XMPP

2011-04-28 Thread randulo

Hi all,

Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org
(formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz)
about Jabber, something the Asterisk community is becoming more
interested in by the day. Join us to learn more about Jabber and SIP
or to share your knowledge and experience. As always, the VUC
discussion includes people from very diverse backgrounds, so it should
be a unique approach to the subject.

All the info to connect is on this page: http://vuc.me

- SIP:200...@login.zipdx.com (g722, g711)
- Skype:vuc.me and ld.vuc.me
- IRC #vuc
- PSTN +15672522286
- iNum +883510012394882
- gtalk voipusersconfere...@gmail.com

During the conference hours, there's a widget to join on the above
page as well as an mp3 stream link.

Join us!

:r

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?


On 29/04/11 2:15 PM, Alec Davis wrote:

Let me try to better describe the test senario that I found, and have been
commited to 1.4svn, 1.6.2svn 1.8svn and trunk.
All aspects need to be thrased out though.

Leave Phone-A 2 new messages, and for this example we only have 2 new
messages.

Now to create the problem - (gaps in the message sequence):


Ah, which explains why I'm not seeing that too - we do attach=yes, 
delete=yes


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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available