Re: [asterisk-users] asterisk-users Digest, Vol 221, Issue 2

2023-01-25 Thread Ron Lockard
Have you tried using the EVAL function?

On Tue, Jan 24, 2023, 7:38 PM 
wrote:

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> Today's Topics:
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>1. Global variables in global variables (Antony Stone)
>
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> --
>
> Message: 1
> Date: Wed, 18 Jan 2023 23:08:48 +0100
> From: Antony Stone 
> To: "Asterisk" 
> Subject: [asterisk-users] Global variables in global variables
> Message-ID: <202301182308.48557.antony.st...@asterisk.open.source.it>
> Content-Type: text/plain;  charset="iso-8859-1"
>
> Hi.
>
> I have a very old dialplan (ie: a dialplan for a very old version of
> Asterisk)
> which I've just transferred to Asterisk 16.28.0
>
> The [globals] section of that dialplan includes:
>
> Kphones=SIP/KC470IP/KSnom870
> Sphones=SIP/SYealinkT38G/SGC610IP
> Allphones=${Kphones}&${Sphones}
>
> In the old system, this results in ${Allphones} containing:
>
> SIP/KC470IP/KSnom870/SYealinkT38G/SGC610IP
>
> I can use this in a dial() command.
>
> On the new system, the variable ${Allphones} ends up containing:
>
> ${Kphones}&${Sphones}
>
> (ie: the unexpanded variable names, not the content of those previously-
> defined variables.)
>
> This fairly obviously does not work in a dial() command.
>
>
> a) is this a deliberate backward incompatiblity at some stage in the
> development of Asterisk?
>
> b) if not, is this a known bug?
>
> c) is there some other way I'm supposed to be doing this now, to be able
> to
> define a global variable including the value of another global variable?
>
> d) if not, is there some workaround?
>
>
> Thanks,
>
>
> Antony.
>
> --
> Most people are aware that the Universe is big.
>
>  - Paul Davies, Professor of Theoretical Physics
>
>Please reply to the
> list;
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Re: [asterisk-users] General Kernel practices on CentOS

2017-12-18 Thread Ron Wheeler
Linux x.y.com 3.10.0-693.5.2.el7.x86_64 #1 SMP Fri Oct 20 20:32:50 UTC 
2017 x86_64 x86_64 x86_64 GNU/Linux

I try to keep up with the latest versions of everything.

Ron

On 15/12/2017 5:59 AM, Olivier wrote:

Hello Ron,
Which kernel do you run Asterisk/Freepbx with ?
Cheers

2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwhee...@artifact-software.com 
<mailto:rwhee...@artifact-software.com>>:


CentOS 7 works well with Asterisk.
Install latest CentOS7 with updates install asterisk

I am running FreePBX on CentOS 7.

Ron

On 14/12/2017 10:38 AM, Olivier wrote:

Hello,

I'm used to install Asterisk on Debian stable platforms.

A customer is asking how I would proceed on a CentOS platform.

After a short research (see [1] as an example), I'm wondering
what are general kernel practices on CentOS regarding Asterisk
and when targeting stability:

- Is it recommended to upgrade kernel version(s) (ie moving from
linux 3.10 to 4.3) just after OS installation ?

Best regards





    -- 
Ron Wheeler

President
Artifact Software Inc
email:rwhee...@artifact-software.com <mailto:rwhee...@artifact-software.com>
skype: ronaldmwheeler
phone:866-970-2435, ext 102 <tel:%28866%29%20970-2435>


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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Any impact on VoIP from loss of Net neutrality

2017-12-17 Thread Ron Wheeler


The article seems to focus on ISPs charging more for VoIP traffic.
This is only a money argument but seems to ignore the fact that my call 
from Montreal, Canada to Palm Springs, California will travel through a 
few networks that have no relationship with me.


It appears that carriers could increase latency on "foreign" VoIP 
traffic which could reduce us to being able to buy good local call 
service but unable to use phones for long distance calls.


Should national and multinational companies be concerned that even their 
ability to use the telephone for internal calls could be impacted by 
this.  It seems clear that the ability to make calls to customers and 
suppliers will become uncertain and potentially vary from cases to case.


Ron

On 16/12/2017 1:05 PM, Eric Klein wrote:

Hi Ron

There was an article back in July looking at what might happen

How does the 2017 Net Neutrality Debate Affect VoIP? ( 
https://voipstudio.com/2017-net-neutrality-debate-affect-voip/ )


In general, anything that allows them to charge more, limit, or 
prioritize can affect VoIP.
There were cases in the past where the carriers would do this via Deep 
Packet Inspection (DPI) to block services that competed with their own 
services. So it is not hard to envision this happening in the future.


That said., the vote was not the end of the story. There is still a 
law suit pending on this topic and Congress is being forced to review 
the decision (and potentially finally create a proper law). ( 
http://uproxx.com/news/senate-democrats-cc-net-neutrality-fight/ )


So it is worth it to contact your Senator and let them know what you 
think they are supposed to be doing in your name.


Eric Klein
COO
Greenfield
Main US +1 805 410 1010
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Message: 1
Date: Fri, 15 Dec 2017 09:42:00 -0500
From: Ron Wheeler <rwhee...@artifact-software.com
<mailto:rwhee...@artifact-software.com>>
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] Any impact on VoIP from loss of Net
        neutrality
Message-ID:
       
<72ebd344-8f94-d9f1-40e5-218536d60...@artifact-software.com
<mailto:72ebd344-8f94-d9f1-40e5-218536d60...@artifact-software.com>>
Content-Type: text/plain; charset=utf-8; format=flowed

Has there been any discussion about the the effect of the changes
in net
neutrality to VoIP service quality.

It seems to me that prioritizing streaming traffic from certain
content
delivery companies could have an impact on the latency for VoIP which
could disrupt phone service.

I found this article
https://voipstudio.com/2017-net-neutrality-debate-affect-voip/
<https://voipstudio.com/2017-net-neutrality-debate-affect-voip/>

It seems to be assuming that VoIP traffic only traverses one
network and
that my trunk provider will be able to charge me more and
guarantee that
my traffic get priority but I am pretty sure that at least some of my
traffic crosses many networks.

Am I way off track?


Ron

--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
<mailto:rwhee...@artifact-software.com>
skype: ronaldmwheeler
phone: 866-970-2435, ext 102





--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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[asterisk-users] Any impact on VoIP from loss of Net neutrality

2017-12-15 Thread Ron Wheeler
Has there been any discussion about the the effect of the changes in net 
neutrality to VoIP service quality.


It seems to me that prioritizing streaming traffic from certain content 
delivery companies could have an impact on the latency for VoIP which 
could disrupt phone service.


I found this article 
https://voipstudio.com/2017-net-neutrality-debate-affect-voip/


It seems to be assuming that VoIP traffic only traverses one network and 
that my trunk provider will be able to charge me more and guarantee that 
my traffic get priority but I am pretty sure that at least some of my 
traffic crosses many networks.


Am I way off track?


Ron

--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] General Kernel practices on CentOS

2017-12-15 Thread Ron Wheeler
It currently runs Linux 3.10.0-693.5.2.el7.x86_64 on x86_64 which I 
believe is the latest CentOS 7.



I apply updates as they are issues by the CentOS team.

I just installed the latest FreePPBX from 
https://www.freepbx.org/downloads/ on bare hardware. This included 
Sangoma's version of Centos 7 build 1701. After the install, I apply 
updates as they are issues by the CentOS team.



Works fine.

Ron


On 15/12/2017 5:59 AM, Olivier wrote:

Hello Ron,
Which kernel do you run Asterisk/Freepbx with ?
Cheers

2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwhee...@artifact-software.com 
<mailto:rwhee...@artifact-software.com>>:


CentOS 7 works well with Asterisk.
Install latest CentOS7 with updates install asterisk

I am running FreePBX on CentOS 7.

Ron

On 14/12/2017 10:38 AM, Olivier wrote:

Hello,

I'm used to install Asterisk on Debian stable platforms.

A customer is asking how I would proceed on a CentOS platform.

After a short research (see [1] as an example), I'm wondering
what are general kernel practices on CentOS regarding Asterisk
and when targeting stability:

- Is it recommended to upgrade kernel version(s) (ie moving from
linux 3.10 to 4.3) just after OS installation ?

Best regards





    -- 
Ron Wheeler

President
Artifact Software Inc
email:rwhee...@artifact-software.com <mailto:rwhee...@artifact-software.com>
skype: ronaldmwheeler
phone:866-970-2435, ext 102 <tel:%28866%29%20970-2435>


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

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--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

-- 
_
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Re: [asterisk-users] General Kernel practices on CentOS

2017-12-14 Thread Ron Wheeler

CentOS 7 works well with Asterisk.
Install latest CentOS7 with updates install asterisk

I am running FreePBX on CentOS 7.

Ron
On 14/12/2017 10:38 AM, Olivier wrote:

Hello,

I'm used to install Asterisk on Debian stable platforms.

A customer is asking how I would proceed on a CentOS platform.

After a short research (see [1] as an example), I'm wondering what are 
general kernel practices on CentOS regarding Asterisk and when 
targeting stability:


- Is it recommended to upgrade kernel version(s) (ie moving from linux 
3.10 to 4.3) just after OS installation ?


Best regards





--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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

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

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Re: [asterisk-users] Rewrite Outgoing Number

2017-12-14 Thread Ron Wheeler

On 14/12/2017 10:36 AM, basti wrote:

On 14.12.2017 16:30, basti wrote:
Hello,
I am new on asterisk and do some tests on freepbx.

I have 2 SIP provider:

Provider1: In-/Out- Flatrate, only 1 Number
Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers

These are trunks.


On Asterisk site i have 3 phones
(branch ??, don't know how its called in asterisk)

These are your extensions


Is it possible to do something like:

Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: Number1/Provider1
Phone 2: Incoming Call: Number1/Provider2 Outgoing Call: Number1/Provider1
Phone 3: Incoming Call: Number2/Provider2 Outgoing Call: Number1/Provider1

I have forgotten an essential thing:

Phone2 und Phone 3 should use  Line Number1/Provider1 for Outgoing Call
but show Number1/Provider2 or Number2/Provider2 on caller side.


I gather that you are talking about Caller ID. You can likely specify 
what you want but you need to look at the caller Id setting on the 
Extension setting form.


It looks like you are trying to tie extensions to trunks directly - No 
press 1 for Ron, 2 for Paul, 3 for tech support, etc.
You want incoming calls to number xxx- to go to Ron's phone and 
yyy-y to go to Paul?

Both should be easy to do and not too hard to set up under FreePBX.





--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

2017-12-12 Thread Ron Wheeler

https://community.freepbx.org

https://www.voip-info.org/wiki/view/Asterisk+Mailing+Lists



On 12/12/2017 11:43 AM, Doug Lytle wrote:

If you do not get an answer from the user list, you might try a post to
the dev list.
It is a bit more active and some of the people watching for dev news
might be able to help you.

Surely, you mean the Biz List

Doug



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

2017-12-12 Thread Ron Wheeler

SLA is just one factor in the decision to select a vendor.

If you do not get an answer from the user list, you might try a post to 
the dev list.
It is a bit more active and some of the people watching for dev news 
might be able to help you.


I used a consultant of the first go around but took over the support myself.
I have a technical background but the FreeBPX user interface is pretty 
easy to set up and administer.

I have to provide my own SLA ;-)

You also have the advantage of a complete duplication of your trunks so 
you can do adequate testing before going live which is a lot better than 
the case where you have to do the switchover and testing over a weekend!


The user group is pretty helpful as well.

Have you tried doing the installation and trunk setup?

You can also use free softphones to test your extensions before you 
actually purchase the phones.
I use the free version of Zoiper on my Android cell as a production 
phone but you can use that to test each extension as you set it up.


The IVR setup is pretty straightforward.

Are there any potential issues that are of particular concern. Ring 
groups, IVR menu design?


Ron

On 12/12/2017 10:30 AM, basti wrote:

I know but this is not my sole decision.

On 12.12.2017 16:17, Ron Wheeler wrote:

If your phone system goes down and you can not get it back up until
tomorrow afternoon because your support person is on another project,
you may wish you had an SLA.



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


--
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Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

2017-12-12 Thread Ron Wheeler


If your phone system goes down and you can not get it back up until 
tomorrow afternoon because your support person is on another project, 
you may wish you had an SLA.


I hope that this extra info helps find a solution.

Ron
On 12/12/2017 3:41 AM, basti wrote:

Size:
- one location
- 15 IP Phones ( 1 dect)
- Create new voip trunk (current are ISDN) (30 number block)
- LTS is important
- an SLA is optional at the moment there is no one


On 11.12.2017 22:31, Ron Wheeler wrote:

You might want to add some details
- size of the project
  -- number of locations
  -- number of extensions
- are you converting your trunks?
- what are your thoughts on hardware - brands, type of stations
- what type of long-term support do you want the consultant to provide?
- what size company do you want to deal with - one man shop with a
genius in charge that you may only be able to reach after hours or a
shop with techs of various skill levels that can give you a believable SLA.

Ron

On 11/12/2017 3:53 PM, basti wrote:

Hello,

we plan to move a PBX to asterisk and searching for Support and a
Phonehardware Reseller in Germany.

The should be no license costs per User / Server.

- Install Configure Asterisk for our specification
- Install FreePBX or similar (optional)
- Resell Hardware

Thanks for any suggest.

Best Regards,
basti




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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

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

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

asterisk-users mailing list
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Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

2017-12-11 Thread Ron Wheeler

You might want to add some details
- size of the project
 -- number of locations
 -- number of extensions
- are you converting your trunks?
- what are your thoughts on hardware - brands, type of stations
- what type of long-term support do you want the consultant to provide?
- what size company do you want to deal with - one man shop with a 
genius in charge that you may only be able to reach after hours or a 
shop with techs of various skill levels that can give you a believable SLA.


Ron

On 11/12/2017 3:53 PM, basti wrote:

Hello,

we plan to move a PBX to asterisk and searching for Support and a
Phonehardware Reseller in Germany.

The should be no license costs per User / Server.

- Install Configure Asterisk for our specification
- Install FreePBX or similar (optional)
- Resell Hardware

Thanks for any suggest.

Best Regards,
basti




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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

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

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

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Re: [asterisk-users] Showing CallerID on multiple phones

2017-12-11 Thread Ron Wheeler

How do you want the last line to work?
Which input choices need to ring on all lines?
- If you select "n" from the IVR menu, do you want to ring everywhere? 
Other selections just go where the IVR menu currently send the call.


Can the option "n" send the call to a ring group? I have never tried 
anything like this.


Perhaps if you clarify this, someone might have a suggestion.

Ron


On 11/12/2017 9:16 AM, Tech Support wrote:


Hello;

    I certainly appreciate your response. In fact, I used that exact 
solution for three of the incoming lines. I setup ring groups and a 
silent ringtone for each phone. Unfortunately, the last incoming line 
is more complicated and uses an IVR with multiple input choices, so 
the solution is not as clear cut as for the other ones. That’s why I 
was trying to look at other options.


Best Regards;

John V.

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

*Sent:* Friday, December 08, 2017 03:06 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Showing CallerID on multiple phones

I could be way off track but have you looked at ring groups.

Have all of the phones ring (maybe a mute or special ring tone, if 
this is possible) so that everyone on the list of extensions sees the 
incoming call.
If no one picks it up by ring x, have it go to another phone or to 
voice mail.


https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/ 
might be useful.


Ron


On 08/12/2017 2:17 PM, Tech Support wrote:

All;

    I have an interesting scenario where I have a small office
with maybe half a dozen phones and several incoming lines. The
calls are routed based on the DID that people call. What they
would like is when a call comes in to a single phone to have all
the phones show the CallerID. That way they can decide if they
should pick up the call or not using call pickup. I’ve been
looking at products such as the one from Camrivox that interfaces
with different CRM packages or Outlook, but I was wondering if a
way was available to show the calls on their phones.

Thanks in Advance;

    John V.



--
Ron Wheeler
President
Artifact Software Inc
email:rwhee...@artifact-software.com <mailto:rwhee...@artifact-software.com>
skype: ronaldmwheeler
phone: 866-970-2435, ext 102



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
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Re: [asterisk-users] Showing CallerID on multiple phones

2017-12-08 Thread Ron Wheeler

I could be way off track but have you looked at ring groups.

Have all of the phones ring (maybe a mute or special ring tone, if this 
is possible) so that everyone on the list of extensions sees the 
incoming call.
If no one picks it up by ring x, have it go to another phone or to voice 
mail.


https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/ 
might be useful.


Ron


On 08/12/2017 2:17 PM, Tech Support wrote:


All;

    I have an interesting scenario where I have a small office with 
maybe half a dozen phones and several incoming lines. The calls are 
routed based on the DID that people call. What they would like is when 
a call comes in to a single phone to have all the phones show the 
CallerID. That way they can decide if they should pick up the call or 
not using call pickup. I’ve been looking at products such as the one 
from Camrivox that interfaces with different CRM packages or Outlook, 
but I was wondering if a way was available to show the calls on their 
phones.


Thanks in Advance;

John V.





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skype: ronaldmwheeler
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Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

2017-03-15 Thread Ron Wheeler

Great.
Let me know how your policy works out.
I would not mind trying it myself.
I have no intrinsic objection to doing things the right way but 
sometimes one just needs to get the phones working!


Ron

On 15/03/2017 4:06 PM, Dan Cropp wrote:


Thank you Jason

After following your steps, Asterisk starts up each time even after 
the reset.


I will look into creating an SELinux policy exception for Asterisk.

Have a great day!

Dan

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

*Sent:* Wednesday, March 15, 2017 1:52 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] Having problem getting Asterisk to 
work on CentOS 7


Dan – you probably installed the init script (look in /etc/init.d for 
an ‘asterisk’ file).  Asterisk includes the older init style scripts 
which are *compatible* with systemd but you don’t have as much control 
compared to creating an Asterisk systemd file.  (SystemD service files 
replace InitD scripts).  So that might be part of the solution, but first…


If disabling Selinux allows Asterisk to run as you expect then you can 
create an selinux policy exception for Asterisk – BUT, ignore that for 
now.  Just keep SElinux disabled (edit /etc/sysconfig/selinux and set 
to disabled) and come back to that later.


So in preparation to diagnose further:

1.Disable asterisk service (systemctl disable asterisk)

2.Disable selinux (as described above)

3.Reboot.

Next, try to start asterisk with ‘systemctl start asterisk’.  Does it 
work as expected?  If no, what user have you logged in with?


If not root, su to root and try again.  Did it asterisk service start 
properly?


If yes, you should create a systemd service file and use the 
‘user=root’ parameter (and remove the initd service script).


Does Asterisk start properly now every time?  If yes re-enable to your 
systemd Asterisk service to start with the system.


I don’t see any attachment (probably stripped by the list manager) but 
that shouldn’t matter – if your Asterisk service is not running as 
root that would explain a range of strange behaviours.


**Jason**

*From:*asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>[mailto:asterisk-users-boun...@lists.digium.com] 
*On Behalf Of *Dan Cropp

*Sent:* Wednesday, March 15, 2017 12:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com>>
*Subject:* Re: [asterisk-users] Having problem getting Asterisk to 
work on CentOS 7


Thanks Jason.

I will try to explain what I’m seeing for this issue.

I did a fresh install of CentOS 7 Minimal into a VM with VMWare 
Workstation.  Followed the Asterisk from Source instructions using 
pjproject 2.6 and asterisk 13.14.0 for the configure, install, …   At 
the end of the asterisk portion, I ran the make config which I 
understand installs the Initialization scripts.


After this, when I restart my CentOS 7 Minimal, I was seeing the 
safe_asterisk process, but asterisk would not start.  I could run it 
from the command line and it would run.


It was suggested that it’s an selinux problem.  They had me try 
‘setenforce 0’.  After this, asterisk process starts running.


As I understand it, there was mention of using systemd instead of 
using safe_asterisk.


Other e-mails indicated I should look at the audit.log, so I included 
that information.  This audit.log mentioned astdb.sqlite3, so I wasn’t 
sure if that’s the problem.


I also just tried a restart and ran ‘systemctl start asterisk’.  This 
did not start the asterisk process.


Through the various recommendations, I’ve become confused on what the 
correct path would be.  I have had zero problems with Debian and 
Asterisk for many years.  Making the change to CentOS.  Followed the 
instructions from asterisk.org, but for some reason I hit a problem 
with this on my CentOS VM.


https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source

Simply looking for guidance on what the correct approach to solve this 
problem is.


Have a great day!

Dan

*From:*asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>[mailto:asterisk-users-boun...@lists.digium.com] 
*On Behalf Of *Telium Technical Support

*Sent:* Wednesday, March 15, 2017 11:08 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] Having problem getting Asterisk to 
work on CentOS 7


The history of the question is lost (in the mail thread) so I’ll jump 
in based on what I could see in my recent mail and the subject line:


-The ASTDB should have no impact on Asterisk service start (which I 
assume is the problem given the subject line)


-If you disabled SElinux then that’s not the problem in starting asterisk

From another posting it appears that you

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

2017-03-15 Thread Ron Wheeler


Have you tried to turn of Selinux permanently?
I don't see why it would be required on a system that has no users.
In /etc/selinux/config I set SELINUX to disabled

# This file controls the state of SELinux on the system.
# SELINUX= can take one of these three values:
# enforcing - SELinux security policy is enforced.
# permissive - SELinux prints warnings instead of enforcing.
# disabled - No SELinux policy is loaded.
SELINUX=disabled
# SELINUXTYPE= can take one of three two values:
# targeted - Targeted processes are protected,
# minimum - Modification of targeted policy. Only selected processes 
are protected.

# mls - Multi Level Security protection.
SELINUXTYPE=targeted

This will make sure that you start up without Selinux.

This may partition your problem so that you know where to focus.

I use systemd for everything.
All of my servers are running some version of CentOS 7

Ron

On 15/03/2017 12:40 PM, Dan Cropp wrote:


Thanks Jason.

I will try to explain what I’m seeing for this issue.

I did a fresh install of CentOS 7 Minimal into a VM with VMWare 
Workstation.  Followed the Asterisk from Source instructions using 
pjproject 2.6 and asterisk 13.14.0 for the configure, install, …   At 
the end of the asterisk portion, I ran the make config which I 
understand installs the Initialization scripts.


After this, when I restart my CentOS 7 Minimal, I was seeing the 
safe_asterisk process, but asterisk would not start.  I could run it 
from the command line and it would run.


It was suggested that it’s an selinux problem.  They had me try 
‘setenforce 0’.  After this, asterisk process starts running.


As I understand it, there was mention of using systemd instead of 
using safe_asterisk.


Other e-mails indicated I should look at the audit.log, so I included 
that information.  This audit.log mentioned astdb.sqlite3, so I wasn’t 
sure if that’s the problem.


I also just tried a restart and ran ‘systemctl start asterisk’.  This 
did not start the asterisk process.


Through the various recommendations, I’ve become confused on what the 
correct path would be.  I have had zero problems with Debian and 
Asterisk for many years.  Making the change to CentOS.  Followed the 
instructions from asterisk.org, but for some reason I hit a problem 
with this on my CentOS VM.


https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source

Simply looking for guidance on what the correct approach to solve this 
problem is.


Have a great day!

Dan

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

*Sent:* Wednesday, March 15, 2017 11:08 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] Having problem getting Asterisk to 
work on CentOS 7


The history of the question is lost (in the mail thread) so I’ll jump 
in based on what I could see in my recent mail and the subject line:


-The ASTDB should have no impact on Asterisk service start (which I 
assume is the problem given the subject line)


-If you disabled SElinux then that’s not the problem in starting asterisk

From another posting it appears that you can start Asterisk from the 
binary, and from safe_asterisk.  If that’s correct, then are you able 
to start/stop Asterisk from the service file? With CentOS7 that would be:


systemctl start asterisk

Is your asterisk service file present?  (You can create one easily 
based on samples on the internet).  If you have an asterisk service 
file but startup fails post the relevant portion of your syslog 
(journalctl).


If your question has changed (you mentioned ‘the first problem’) then 
ignore the above; jumping in late.


**Jason**





--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

2017-03-15 Thread Ron Wheeler
" subj=system_u:system_r:asterisk_t:s0 
key=(null)
type=AVC msg=audit(1489588781.633:1177): avc:  denied  { getattr } for  pid=3851 comm="asterisk" 
path="/var/lib/asterisk/astdb.sqlite3" dev="dm-0" ino=100884225 
scontext=system_u:system_r:asterisk_t:s0 tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588781.633:1177): arch=c03e syscall=4 success=no exit=-13 a0=27cf470 
a1=7a2517d0 a2=7a2517d0 a3=7a2514f0 items=0 ppid=1485 pid=3851 auid=4294967295 uid=0 
gid=0 euid=0 suid=0 fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 
comm="asterisk" exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 
key=(null)
type=AVC msg=audit(1489588781.633:1178): avc:  denied  { read write } for  pid=3851 comm="asterisk" 
name="astdb.sqlite3" dev="dm-0" ino=100884225 scontext=system_u:system_r:asterisk_t:s0 
tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588781.633:1178): arch=c03e syscall=2 success=no exit=-13 a0=27cf470 
a1=80042 a2=1a4 a3=7a251420 items=0 ppid=1485 pid=3851 auid=4294967295 uid=0 gid=0 euid=0 
suid=0 fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 comm="asterisk" 
exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 key=(null)
type=AVC msg=audit(1489588781.633:1179): avc:  denied  { read } for  pid=3851 comm="asterisk" 
name="astdb.sqlite3" dev="dm-0" ino=100884225 scontext=system_u:system_r:asterisk_t:s0 
tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588781.633:1179): arch=c03e syscall=2 success=no exit=-13 a0=27cf470 
a1=8 a2=1a4 a3=27cf470 items=0 ppid=1485 pid=3851 auid=4294967295 uid=0 gid=0 euid=0 suid=0 
fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 comm="asterisk" 
exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 key=(null)
type=AVC msg=audit(1489588785.830:1180): avc:  denied  { getattr } for  pid=3857 comm="asterisk" 
path="/var/lib/asterisk/astdb.sqlite3" dev="dm-0" ino=100884225 
scontext=system_u:system_r:asterisk_t:s0 tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588785.830:1180): arch=c03e syscall=4 success=no exit=-13 
a0=7ffd6605ff40 a1=7ffd6605ff80 a2=7ffd6605ff80 a3=8913bc items=0 ppid=1485 pid=3857 
auid=4294967295 uid=0 gid=0 euid=0 suid=0 fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 
comm="asterisk" exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 
key=(null)
type=AVC msg=audit(1489588785.834:1181): avc:  denied  { getattr } for  pid=3857 comm="asterisk" 
path="/var/lib/asterisk/astdb.sqlite3" dev="dm-0" ino=100884225 
scontext=system_u:system_r:asterisk_t:s0 tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588785.834:1181): arch=c03e syscall=4 success=no exit=-13 a0=1be0de0 
a1=7ffd6605f890 a2=7ffd6605f890 a3=7ffd6605f5b0 items=0 ppid=1485 pid=3857 auid=4294967295 uid=0 
gid=0 euid=0 suid=0 fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 
comm="asterisk" exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 
key=(null)
type=AVC msg=audit(1489588785.834:1182): avc:  denied  { read write } for  pid=3857 comm="asterisk" 
name="astdb.sqlite3" dev="dm-0" ino=100884225 scontext=system_u:system_r:asterisk_t:s0 
tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588785.834:1182): arch=c03e syscall=2 success=no exit=-13 a0=1be0de0 
a1=80042 a2=1a4 a3=7ffd6605f4e0 items=0 ppid=1485 pid=3857 auid=4294967295 uid=0 gid=0 euid=0 
suid=0 fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 comm="asterisk" 
exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 key=(null)
type=AVC msg=audit(1489588785.834:1183): avc:  denied  { read } for  pid=3857 comm="asterisk" 
name="astdb.sqlite3" dev="dm-0" ino=100884225 scontext=system_u:system_r:asterisk_t:s0 
tcontext=unconfined_u:object_r:var_lib_t:s0 tclass=file
type=SYSCALL msg=audit(1489588785.834:1183): arch=c03e syscall=2 success=no exit=-13 a0=1be0de0 
a1=8 a2=1a4 a3=1be0de0 items=0 ppid=1485 pid=3857 auid=4294967295 uid=0 gid=0 euid=0 suid=0 
fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 comm="asterisk" 
exe="/usr/sbin/asterisk" subj=system_u:system_r:asterisk_t:s0 key=(null)


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, March 15, 2017 3:29 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 
7

On Tue, Mar 14, 2017 at 02:46:19PM -0400, R

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

2017-03-14 Thread Ron Wheeler

https://docs.fedoraproject.org/en-US/Fedora/11/html/Security-Enhanced_Linux/sect-Security-Enhanced_Linux-Working_with_SELinux-Enabling_and_Disabling_SELinux.html

If disabling Selinux solves your problem, then your problem may be 
related to Selinux.

If it does not change yout problem, you may want to look elsewhere.

It seems that a lot of  things do not work with Selinux or 
have no instructions about how to make them work with Selinux that it 
almost seems like a useless feature.


Ron

On 14/03/2017 2:21 PM, Tzafrir Cohen wrote:

On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis wrote:

Hello,

Did you disable selinux ? It usually causes troubles when starting asterisk
as a service. You can do this with : setenforce 0 (this will not totally
disable selinux, but switch it to a permissive mode).

Generally before advising that, check if this is the error:

   tail -f /var/log/audit/audit.log

and try the command.

Is there any open bug for a security policy for Asterisk?




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

2017-03-14 Thread Ron Wheeler

I have  FreePBX 14.0.1beta20 running on Centos 7.3.

What problems are you having?
The latest emails don't have any details about the problem or what you 
have tried.


Ron

On 14/03/2017 2:21 PM, Tzafrir Cohen wrote:

On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis wrote:

Hello,

Did you disable selinux ? It usually causes troubles when starting asterisk
as a service. You can do this with : setenforce 0 (this will not totally
disable selinux, but switch it to a permissive mode).

Generally before advising that, check if this is the error:

   tail -f /var/log/audit/audit.log

and try the command.

Is there any open bug for a security policy for Asterisk?




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Ron Wheeler

CentOS 7 uses firewalld to control TCP amd UDP access.

The iptables configuration will be overwritten and dynamically changed 
by Firewalld so don't count on the old practice of manipulating iptables 
directly.


I recently moved our Asterisk from an old CentOS to CentOS 7 running 
FreePBX 14.0.1.beta2.


You can add a firewalld service yp /etc/firewalld/services like mine.
[root@firewall0 services]# cat Asterisk.xml


  asterisk
  Asterisk PBX
  
  
  
  
  


You then permit this service in your interface (zones) as a service
 

I also added a rule to get some logging on the Asterisk ports while 
getting things up and running.

  



  
  


I did this all on my exterior firewall which is also a CentOS 7 system.
On the Asterisk server, I do not block anything which is not a best 
practice but the entire internal network is very small and I consider it 
to be secure.


You (and I) should control the interface using Firewalld with the same 
service and zone specifications.






On 30/01/2017 12:13 PM, Motty Cruz wrote:

I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1


On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from

here:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar
.gz


I continue to see errors like this:
[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:

Retransmission timeout reached on transmission
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>>

Firewall?

Doug




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk Behind Firewall

2016-01-04 Thread Ron Wheeler

Both work.
If you have enough IP addresses to dedicate one to your Asterisk server, 
that removes one node in the path from the world.
You will need a firewall on the Asterisk server to protect it from 
outside meddling.
If you can put the Asterisk server on the same network as the SIP 
devices (using a second NIC) that should help performance.


Is the SIP network on the same network as your internet/data LAN?

Ron

On 04/01/2016 1:15 PM, IPN Comm wrote:
I was wondering if anyone can give me any pointers or insights of 
whether or not to have an asterisk server behind a firewall.


I have always ran Asterisk on a public IP but was wondering if I 
should move it to a local IP behind a firewall.


I am looking to set up a location with 300 SIP phones.

Normally, I would put the Asterisk server on one public IP and let the 
SIP phones get DHCP from a router on a different IP and they would 
register to the Public Asterisk server from that IP address.


Should I move the asterisk server behind the same router?

If so, how should the server be set up and what is the best 
router/firewall hardware to accomplish this environment?


Thanks,
-H





--
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President
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Why no CentOS 7 repos?

2015-07-27 Thread Ron Wheeler

On 27/07/2015 1:51 PM, Steve Edwards wrote:

Any particular reason CentOS 7 repos aren't available?

I'm finding integration issues with CentOS 6's ancient versions of 
MySQL and PHP with third party applications.



You might have o upgrade MySQL and PHP outside of the Centos distribution.
I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11.

Ron

--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Why no CentOS 7 repos?

2015-07-27 Thread Ron Wheeler

On 27/07/2015 2:38 PM, Steve Edwards wrote:

On 27/07/2015 1:51 PM, Steve Edwards wrote:



Any particular reason CentOS 7 repos aren't available?

I'm finding integration issues with CentOS 6's ancient versions of 
MySQL and PHP with third party applications.


On Mon, 27 Jul 2015, Ron Wheeler wrote:

You might have o upgrade MySQL and PHP outside of the Centos 
distribution. I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with 
FreePBX 2.11.


I really prefer to keep to the repos. It's a numbers thing:

) I don't want to spend the time (aka $$$) to track patches to packages.

) I don't want to be 'different.' I want to run the same versions as 
others so I don't get to discover and resolve incompatibilities all by 
myself.


CentOS 7 was released over a year ago. Seems overdue to me.



I absolutely agree with your reasoning but sometimes the realities of 
using open source means that some of the money that you save on 
licensing has to be spent on support or worse (pioneering).


If you use PHP and MySQL from a repo, Yum will track and install updates.
https://dev.mysql.com/downloads/repo/yum/
https://webtatic.com/projects/yum-repository/

I am not sure how you can be sure to match up with others since everyone 
has their own tolerance to change and will often let a sleeping dog lie 
until it bites them.
Your Centos 6 (or 7) will have much more up to date versions of MySQL 
and PHP that mine since mine was running in a stable configuration of 
OS, PHP and database over a year ago.


I only discovered the MySQL and PHP repos after I had a stable Asterisk 
(which has not bit bitten me yet) so I have only used these on Centos 7 
for other servers not related to Asterisk.


I hope that this helps.

Ron

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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] switching from SIP to Skype..or not

2015-03-13 Thread Ron Wheeler


Sorry for the empty message. Pressed the wrong button.


I have been wrestling with a pretty generic Asterisk configuration 
(version 11.11.0 ) set up with FreePBX.

The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled.
I was using Eyebeam and am now trying Jitsi. Jitsi has a number of 
codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and 
telephone-event
The internet connection from the workstation to my internet supplier 
(workstation to firewall/router to speed test server at ISP) tests at 
13MBs incoming 6Mbs outgoing.


The problem has always been great sound from the other telephone and 
choppy sound (dropped sound fragments) from me to the caller with only 
one call going through Asterisk and the network pretty much dedicated to 
the my workstation.


This has survived upgrades of everything (firewall, Asterisk server, 
workstation)


This has reduced my Asterisk telephone to an answering machine with 
Skype as my way of actually talking to people.
This fixes the sound issues and is actually cheaper since I pay a low 
monthly fixed cost for Skype access to all North American telephones.
Skype does not seem to have an problem traversing the same network even 
with two way video active or during multi-party conferences (mix of 
Skype and telephones in the group).


I would like to have a reliable 2 way conversation using Asterisk but 
have not found any suggestions about the source of the problem or how to 
fix it.


Ron

On 12/03/2015 10:21 AM, Bryant Zimmerman wrote:

Hey all
We have been working with SIP for years. It has the potential to be 
better than Skype. It is really all in the implementation.
Not all SIP soft clients are equal nor are the networks and computers 
they are running on.
I will not bash Skype. We have tested it and in most cases choose not 
to use it. It has it's place and is good for the user that meets it's 
specific target demographic.  SIP is a sold communications protocol 
that can communication with codecs of differ audio and video quality 
levels, and supports industry standard software and hardware endpoints.
With SIP you get to choose how good your quality is. With Skype 
Microsoft does.
It comes down to what do you want to achieve, how much resource do you 
want to put in to it, and are you committed to a bit more work for a 
lot more options and better quality, or do you want a quick and easy 
solution with differing limits. Both solutions have their place.  To 
me SIP vs Skype is like complaining apples and carrots do you want 
fruit or veggies you get to choose.
You can choose to agree or disagree with my statements. I hope they 
are useful to some.

Thanks

Bryant

*From*: Ron Wheeler rwhee...@artifact-software.com
*Sent*: Thursday, March 12, 2015 9:40 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] switching from SIP to Skype..or not
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.

I have SIP softphone with Asterisk server and Skype on the same
workstation.
Skype just works better over the same network.

Ron

On 12/03/2015 9:26 AM, A J Stiles wrote:
 On Thursday 12 Mar 2015, Thufir wrote:
 I'm testing Asterisk at home, crummy connection. Skype works fine for
 me, but every SIP client, even without using Asterisk, fails to 
connect.

 That's ok.

 Is swapping out SIP for Skype a big deal?
 Stay away from Skype! It is a toxic, proprietary product. The lack of
 interoperability by design is the antithesis of what a telecommunication
 system should be about -- and the extent to which they have gone to 
thwart any

 attempt at interoperability is truly shocking.

 For connecting two Asterisk installations to each other over the 
Internet, IAX

 is better than SIP -- that's what it was designed for.



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] switching from SIP to Skype..or not

2015-03-13 Thread Ron Wheeler

On 12/03/2015 10:21 AM, Bryant Zimmerman wrote:

Hey all
We have been working with SIP for years. It has the potential to be 
better than Skype. It is really all in the implementation.
Not all SIP soft clients are equal nor are the networks and computers 
they are running on.
I will not bash Skype. We have tested it and in most cases choose not 
to use it. It has it's place and is good for the user that meets it's 
specific target demographic.  SIP is a sold communications protocol 
that can communication with codecs of differ audio and video quality 
levels, and supports industry standard software and hardware endpoints.
With SIP you get to choose how good your quality is. With Skype 
Microsoft does.
It comes down to what do you want to achieve, how much resource do you 
want to put in to it, and are you committed to a bit more work for a 
lot more options and better quality, or do you want a quick and easy 
solution with differing limits. Both solutions have their place.  To 
me SIP vs Skype is like complaining apples and carrots do you want 
fruit or veggies you get to choose.
You can choose to agree or disagree with my statements. I hope they 
are useful to some.

Thanks

Bryant

*From*: Ron Wheeler rwhee...@artifact-software.com
*Sent*: Thursday, March 12, 2015 9:40 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] switching from SIP to Skype..or not
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.

I have SIP softphone with Asterisk server and Skype on the same
workstation.
Skype just works better over the same network.

Ron

On 12/03/2015 9:26 AM, A J Stiles wrote:
 On Thursday 12 Mar 2015, Thufir wrote:
 I'm testing Asterisk at home, crummy connection. Skype works fine for
 me, but every SIP client, even without using Asterisk, fails to 
connect.

 That's ok.

 Is swapping out SIP for Skype a big deal?
 Stay away from Skype! It is a toxic, proprietary product. The lack of
 interoperability by design is the antithesis of what a telecommunication
 system should be about -- and the extent to which they have gone to 
thwart any

 attempt at interoperability is truly shocking.

 For connecting two Asterisk installations to each other over the 
Internet, IAX

 is better than SIP -- that's what it was designed for.



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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phone: 866-970-2435, ext 102

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Re: [asterisk-users] switching from SIP to Skype..or not

2015-03-12 Thread Ron Wheeler
Your characterization may be true but Skype works much better than SIP 
when it comes to sound quality.


I have SIP softphone with Asterisk server and Skype on the same 
workstation.

Skype just works better over the same network.

Ron

On 12/03/2015 9:26 AM, A J Stiles wrote:

On Thursday 12 Mar 2015, Thufir wrote:

I'm testing Asterisk at home, crummy connection.  Skype works fine for
me, but every SIP client, even without using Asterisk, fails to connect.
That's ok.

Is swapping out SIP for Skype a big deal?

Stay away from Skype!  It is a toxic, proprietary product.  The lack of
interoperability by design is the antithesis of what a telecommunication
system should be about -- and the extent to which they have gone to thwart any
attempt at interoperability is truly shocking.

For connecting two Asterisk installations to each other over the Internet, IAX
is better than SIP -- that's what it was designed for.




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Problems with the voice quality under load

2015-03-02 Thread Ron Wheeler

Have you done the math for the network connections? BTF and external
What bit rates for the sound?
What codecs?
How are calls coming in - SIP - analogue

Disks OK(low IO per second)? Caching working OK?

CPU may not be the problem if your CPU utilization is really that low.

Ron

On 02/03/2015 10:26 AM, Mordechay Kaganer wrote:

B.H.

Hello, all :-)

We have a cluster of Asterisk (v. 11.9) servers that host IVR 
applications. The servers work behind SIP proxy (kamailio) for load 
balancing.


All servers are in 2 processor configuration, 8-10 cores per CPU.

When a particular server gets about 500 concurrent calls, the sound 
quality begins to degrade, the sound plays slowly and with clicks. As 
far as i understand, it's because asterisk is unable to send the voice 
stream in time i.e. the server is overloaded.


What i don't understand is, at the time that the server appears to be 
overloaded and the audio quality is bad, actual server's load is no 
more than 30-40% (60-70% idle CPU on average). IMHO, this indicates 
that for some reason the server is unable to use it's CPU capacity 
efficiently. May be because of some kind of thread contention inside 
asterisk?


I have read blogs that advice to divide physical server into several 
VMs and they claim that this will improve the total capacity. In my 
own experience, this did not work very well and seems like the 
visualization actually made the quality worse.


Do you have any advice for me (other than purchasing more servers ;-) ?

Thanks!

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--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Is Asterisk a Linux only system?

2015-02-12 Thread Ron Wheeler

Why not just bite the bullet and move to a supported Linux?
- you can be assured that it works
- updates are tested
- help and support is readily available.
- only takes a few minutes to install the whole setup
- configuration should port easily.

There is almost no Linux administration required once it is set up so 
getting deep into the actual OS is not required.

I have used CentOS (5.x and 6.x) for years without any problems.
I have not tried it with CentOS 7 and would recommend sticking with the 
latest CentOS 6  for a while yet.
I am converting the rest of my datacenter to 7 starting with the main 
firewall/router and a virtual host.

Firewall is now in production but it was a bit of a learning curve for me.

There are big differences between 6 and 7 and I would let some other 
Asterisk users switch before going to 7.


Free advice and worth every cent!

Ron

On 12/02/2015 9:25 AM, D'Arcy J.M. Cain wrote:

I know that it runs on other systems but do other ports get the same
attention?  I have been running it on a NetBSD server for about a year
now and while it mostly works it just crashes from time to time with no
explanation or core dump.

I have improved the situation by expanding my intrusion detection but
it still stops every few days or so.  I have a cron job that tests for
it and restarts it when necessary.

Anyone else have experience on non-Linux systems?

Cheers.




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

2014-11-22 Thread Ron Wheeler

You might check your phones as well.
We had this problem early on with a softphone and it was a setting in 
the phone that was set to hang up after 30 seconds of inactivity in 
case of network disruption. For some reason it was detecting network 
disruption in every call even when the calls were proceeding normally.

Unchecking this box solved the problem.

It may not be related to your problem but if it is the cause, you will 
spend a lot of time trying to fix this in Asterisk. :-D At least I did!


On the bright side, it does force people to get point in a hurry!

Ron

On 22/11/2014 12:50 PM, Eric Wieling wrote:

Try setting directmedia=no in sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

but as soon as I configure another sip registration on another server,
outgoing
calls  drop after 32 seconds.

Are both your servers behind the same NAT router?


thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they
interfere at all?

thanks,
yves

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Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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_
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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-08 Thread Ron Wheeler
This is free software supported by volunteers who are helping you 
because somewhere in the past someone helped them or perhaps they were 
just brought up properly.


No one really cares about how many clients you may have. No one here 
will make any money from that.


Have you read the free books on Asterisk?
Have you at least followed the installation instructions to some point 
where you are stuck.


I don't see where you have explained what you want to do with the PBX 
once you have it set up. There are a lot of different applications 
possible - Small office, enterprise with many branches, call center with 
thousands of incoming or outgoing calls, etc.


Have you looked at what phones you want to support - brand, features?

Have you selected a type of trunk that your telephone company offers? Do 
they have a document describing how to interface an Asterisk PBX to 
their trunks?


What were you expecting a total stranger to do for you when you asked 
your question?
If you want to hire a consultant to set up your first few clients and 
train you, than you should say so.
Free advice comes in many forms but there is a limit about how much free 
time each one of us has and you will get different people helping you 
depending on the question that you ask. We are not all experts at 
everything. Most of us are people like you or end-users who are 
supporting their own company's phone system.


Ron


On 08/10/2014 12:34 AM, Dania Asi wrote:


Dear Mr. Adam,

Thank you for you kind words and for judging me.

I am a system integrator and I have a whale clients in UAE , I will 
not proceed further in dealing with Asterisk because of the lack of 
support and because of the rude emails.


I have no idea what is wrong with you people. And I hope you get well 
soon from whatever is happening to you.


*Best Wishes,*

**

*Dania Abu Asi*

Sales Executive Engineer


*Future Trends Establishment***

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

*From:* Adam Goldberg [mailto:a...@agp-llc.com]
*Sent:* Tuesday, October 7, 2014 8:51 PM
*To:* Dania Asi
*Subject:* FW: [asterisk-users] Asterisk Phone ( Telecom feature )



I suggest that your question amounts to please do my homework for 
me.  This may be understandable given that you are a recent grad and 
probably don't have much experience in business communication and/or 
Asterisk complexities.


You cannot expect a mailing list to rush to answer vague, unanswerable 
questions -- nor emails that don't show that you've tried to answer 
the question first.


Consider, if I asked:

I don't understand how to set up Asterisk.  Can 
someone tell me how to do that?


vs.

I have a Dell R210-II with 32g of memory and two gigabit ethernet 
interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and 
have an assortment of Polycom and Snom IP phones. I've configured 
paging as described in 
http://wiki.snom.com/Interoperability/PBX/Asterisk and 
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is 
working for the Snom phones but not the Polycom phones.  Can someone 
point me at what it's going to take to make the Polycom phones work?


I'd expect to get attempts at an answer to the second one, but would 
expect snide and rude comments (at best) to the first one.


Adam Goldberg

AGP, LLC

+1-202-507-9900

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly

Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

JG confirmed that it is possible, but it has not been defined.

Without knowing what kind of instruments you are using, a possible it

would be for a party to dial a 4-digit extension number to talk to 
someone internally, completing a call without using the PRI trunks.


  --Don

-Original Message-

From: asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com


[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

Sent: Tuesday, October 07, 2014 3:41 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: 'Irene Galera'; 'Maysara Orabi'; moham...@futuretrendsest.com 
mailto:moham...@futuretrendsest.com


Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

Dear Mr. Mitual,

Kindly check the attached mail where Mr. JG confirmed to me that is 
possible and I already informed my client of that.


Dear Mr. Steve,

I am not expecting a mailing list to do any work for me. All I was 
asking is for you to guide me because this is the first time we deal 
with Asterisk phones.


Best Wishes,

Dania Abu Asi

Sales Executive Engineer

Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

-Original Message

Re: [asterisk-users] Limit Asterisk

2014-07-23 Thread Ron Wheeler

I would also do some math on the bandwidth requirement.

If you divide your disk bandwidth by your recording bit rate what is the 
theoretical maximum number of calls that you can record at once? Assumes 
that you have infinite CPU and memory and that you can actually drive 
the disks at their maximum.
If this comes out to 300, you are already there. If it comes out to 
3000, you have something wrong in your setup or your assumptions and a 
target to work towards.


What quality are you using in the recording? 44k per second(CD quality 
sound)  uses a lot more bandwidth than 3K (telephone quality)

What encoding are you using?
How low a bit rate can you use and still have usable recordings? If they 
are for legal or audit use, you can go pretty low. If you are recording 
soundtracks for reuse in training or publication, you may require higher 
bit rates.


If you disable recording, how many simultaneous calls can you support? 
Just to be sure that recording is the issue.


Ron

On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
Your bottleneck is most likely your drive bandwidth.  Even with SAS 
drives, you'll need to move to a raid 5+ solution with 6+ drives to 
continue to increase the concurrent calls, or use a storage appliance.


To confirm this, install the tool nmon and use the v and d options to 
bring up the resource usage indicators and drive busy/throughput 
statistics.




On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones 
edua...@ypytecnologia.com.br mailto:edua...@ypytecnologia.com.br 
wrote:


people

I have a running Asterisk 1.8.28 in great Dell server with two
xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This
server is recording all calls (placed to record the audio in a ram
disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each
call runs some validation and AGI's have an auto dialer system
that generates calls over the manager. Calls originate and
terminate via SIP (no transcode).

With this structure, even being a great server, we can not spend
150 simultaneous calls. When it reaches 140, the load average goes
up a lot and the calls start to get very bad audio, tear, etc..
Using the top we see that all the processing is for asterisk. In
this scenario, I think there is some limitation in Asterisk, or
even the manager due to the auto dialer.

Can anyone give me any tips where I can look where is the
bottleneck? I need to get at least 250 calls that server quality.

tks


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Check us out at: http://digium.com · http://asterisk.org





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Re: [asterisk-users] Limit Asterisk

2014-07-23 Thread Ron Wheeler

Do the calculations for both and see what the answer is.
The nice thing about having a model is that you can test configurations 
without actually having to build one until you are confident that it 
should work.


Ron


On 23/07/2014 5:04 PM, Eduardo Leones wrote:

Thanks for the feedback.

In this case SSD disks you think it solves?




2014-07-23 17:29 GMT-03:00 Scott Griepentrog sgriepent...@digium.com 
mailto:sgriepent...@digium.com:


Your bottleneck is most likely your drive bandwidth.  Even with
SAS drives, you'll need to move to a raid 5+ solution with 6+
drives to continue to increase the concurrent calls, or use a
storage appliance.

To confirm this, install the tool nmon and use the v and d options
to bring up the resource usage indicators and drive
busy/throughput statistics.



On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones
edua...@ypytecnologia.com.br
mailto:edua...@ypytecnologia.com.br wrote:

people

I have a running Asterisk 1.8.28 in great Dell server with two
xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This
server is recording all calls (placed to record the audio in a
ram disk), the entire CDR goes straight to MySQL by
cdr_mysql.so. Each call runs some validation and AGI's have an
auto dialer system that generates calls over the manager.
Calls originate and terminate via SIP (no transcode).

With this structure, even being a great server, we can not
spend 150 simultaneous calls. When it reaches 140, the load
average goes up a lot and the calls start to get very bad
audio, tear, etc.. Using the top we see that all the
processing is for asterisk. In this scenario, I think there is
some limitation in Asterisk, or even the manager due to the
auto dialer.

Can anyone give me any tips where I can look where is the
bottleneck? I need to get at least 250 calls that server quality.

tks


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Re: [asterisk-users] Attack on Sip server.

2014-06-27 Thread Ron Wheeler

+1 fail2ban
Very easy and very effective.
On 27/06/2014 10:52 AM, Anurag Rana wrote:

Both Rules* (typo in last mail)


On Fri, Jun 27, 2014 at 8:19 PM, Anurag Rana 
anuragrana31...@gmail.com mailto:anuragrana31...@gmail.com wrote:


I added bot rules TCP as well as UDP.  Still not working.

How changing SIP listen port will prevent it. Please explain.

I will try fail2band.


On Fri, Jun 27, 2014 at 8:16 PM, Prakash N prakas...@tevatel.com
mailto:prakas...@tevatel.com wrote:

Hi,

Install fail2band and change sip listen port to avoid attack

With regards

N.Prakash

From: Anurag Rana mailto:anuragrana31...@gmail.com
Sent: ?27-?06-?2014 08:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] Attack on Sip server.


Hi All.

Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop
it because I am unable to detect the IP address.
I used wireshark to capture the packets.

Although I am using very strong password for my SIP users but
still is there any way to drop these packets and stop this attack.

I tried dropping packet after matching some string (most of
the packets from attacker contains string
'VaxSIPUserAgent/3.1' ) but it failed. Packets are still
flowing in.

iptables -I INPUT 1 -p tcp --dport 5060 -m string --string 
VaxSIPUserAgent --algo bm -j DROP


Its something like this

Registration from '30 sp:30@my_public_ip:5060 failed for
'192.168.xxx.xxx:6373' - Wrong Password

and there are approx 10 request per minute of this type.

Please suggest some way to stop this.


-- 
Anurag Rana

http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a
saintly life in the midst of these materialistic turbulences.





-- 
Anurag Rana

http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a
saintly life in the midst of these materialistic turbulences.





--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly 
life in the midst of these materialistic turbulences.








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Re: [asterisk-users] Need to hire recordings for an IVR

2014-04-07 Thread Ron Wheeler

On 07/04/2014 11:29 AM, CDR wrote:

I wonder if anybody know how to hire Alice or some professional
voice-artist. I need to record 12 messages for a customer.

We have had good success with a local sound studio that uses radio 
personalities for recording.

I like radio announcers for the following:
- good quality
- fast turnaround - can read and understand a script and get it right 
the first time

- ability to find the talent again if you need re-recording.
- neutral accent

Ron

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Re: [asterisk-users] Replying to Posts

2014-03-14 Thread Ron Wheeler
If the sequence was really important, we would just move this whole list 
to a LinkedIn group and have a much better environment for following 
threads and managing the profiles of the group's members.
This is a pretty old system of managing peer support but it always takes 
a while to get legacy systems replaced.


Ron

On 14/03/2014 9:52 AM, James B. Byrne wrote:

On Thu, March 13, 2014 15:32, Kevin Larsen wrote:

On 13/3/14 6:27 pm, Eric Wieling wrote:

This is an example of why I top post.   Who wrote what?

+1-1 = 0

I do not care about where people put their replies so long as I can figure out
who is answering what.  What I do not like to read is this interminable
religious dogma about the 'natural' order of writing.  This is the second or
third list this week in which this B.S. has shown up in my inbox.

In written business communication, in contrast to tech-speak customarily found
on mailing lists, ones answer always goes before any quoted context.  Not
because it has to, it is just that I have seldom, if ever, seen it done any
other way. And regular business communication with non-technical folk
comprises well over 75% of my daily written communication.

And while I understand the cultural motivation behind the dogma of bottom
posting I remain sceptical respecting its utility.  Is there any objective
evidence whatsoever that top or bottom posting makes any difference to the
reader's understanding of the message?  Does any rigorously determined data
exist to support that contention?  If not then this is simply a matter of
trying to impose a set of arbitrary cultural values cloaked in the guise of
technical superiority.



Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.


Outlook can quote correctly, but it is an all or nothing setting it would
appear. Lotus Notes actually handles it better as there is a Reply option
for normal email and a Reply With Internet-Style History that I use for
this list. I don't have any problems following the rules of the list, but
I am fully on the side of the Replies should go at the top group and
would vote for a change in the rules.



And do not even start on the Chevy vs. Ford debate respecting the technical
superiority of Pine over Outlook.  GAWD... Life its too short as it is.





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Re: [asterisk-users] Replying to Posts

2014-03-13 Thread Ron Wheeler

-1
Prefer top posting.
Easy to see if I want to scroll down to see if it is something 
interesting to me.

I get a lot of e-mails each day and scrolling wastes too much time.

But if you have a solution to a problem that I raise, please feel free 
to post it anywhere you like.



On 13/03/2014 11:33 AM, A J Stiles wrote:

(If you want to reply to this message, this is not where your reply goes)


Please, for the benefit of anyone reading the archives in search of answers to
a question, when replying to messages on this list, can everyone try to follow
the natural flow of conversation?  That is, position your reply *AFTER* the
thing you are replying to, not before it.  You may remove quoted material in
order to keep the message size down, but please leave enough of it to preserve
context.

(If you want to reply to a point made in the preceding paragraph, this is
where your reply goes)


If you need to make a point-by-point argument, split up your reply --
inserting artificial paragraph breaks into the quoted material, if necessary --
so each section of your reply follows the point it is addressing.

(If you want to reply to a point made in the preceding paragraph, or the
message as a whole, this is where your reply goes)





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Re: [asterisk-users] Replying to Posts

2014-03-13 Thread Ron Wheeler

On 13/03/2014 9:32 PM, Don Kelly wrote:


 On 13/3/14 6:27 pm, Eric Wieling wrote:
  This is an example of why I top post.   Who wrote what?

 Of course, if you use a mail client that's capable of quoting correctly,
 it all works beautifully.


Kevin Larson sez:

Outlook can quote correctly, but it is an all or nothing setting it 
would appear. Lotus Notes actually handles it better as there is a 
Reply option for normal email and a Reply With Internet-Style History 
that I use for this list. I don't have any problems following the 
rules of the list, but I am fully on the side of the Replies should 
go at the top group and would vote for a change in the rules.


I'll vote again for top posting, and expect my vote to be recognized 
internationally about as much as the Crimean referendum.




But the Russians will get to keep Crimea so don't worry too much about 
our preference for top posting.

In the long run.


--Don






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Re: [asterisk-users] Is this list dead? Or the project?

2014-03-02 Thread Ron Wheeler
It works, there is documentation and you can Google most of the things 
that you are going to run into, if you don't want to read the docs.

The GUI interfaces mostly work if you want to support analog, SIP and VoIP.

Only the odd little issues or people trying to support odd 
configurations result in forum discussions.


Ron

On 02/03/2014 7:44 PM, Doug Lytle wrote:

Stefan Gofferje wrote:

Now it's
hardly 50 new mails per week.
Is the list dead? Or is the project dead?

It's called being a mature project.  And, I don't call averaging 400
messages a month as being a dead list.

And, once I've got several stable systems in production, I don't mess
with them much.


Doug





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Re: [asterisk-users] Asterisk Not Starting after YUM Update

2014-02-12 Thread Ron Wheeler

DAHDI might be the culprit.
You may have had a better version from Asterisk than the new one that 
YUM got you.


Check to see if YUM gave you a new DAHDI. Who's your daddy now?

You may want to rebuild the Asterisk DAHDI and install it over the DAHDI 
from your Linux distro.



Ron

On 12/02/2014 5:22 PM, Tzafrir Cohen wrote:

On Wed, Feb 12, 2014 at 10:44:42PM +0100, Aldo Bergamini wrote:

Hi List,

it feels silly, but here I am.

My Asterisk box is useless, after running a long delayed yum update (Centos 
box).

[snip]


Starting Asterisk very verbosely seems to load the dialplan, but at some point 
I get a segmentation fault. This is new to me!

[…] edited […]
  chan_agent.so = (Agent Proxy Channel)
   == Registered custom function 'EXTENSION_STATE'
  func_extstate.so = (Gets an extension's state in the dialplan)
   == Registered application 'DAHDIBarge'
  app_dahdibarge.so = (Barge in on DAHDI channel application)
   == Registered custom function 'CALLERPRES'
   == Registered custom function 'CALLERID'
  func_callerid.so = (Caller ID related dialplan functions)
[2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A 
transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc.
[2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module 
is supplied under a commercial license granted by Digium, Inc.
[2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see 
the full license text supplied by the accompanying
[2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register 
utility, or ask for a copy from Digium.
Segmentation fault



The problem seems to come after the callerid module loads: does this make sense?

BTW: I do have a G729 pack of licenses (they were actually active without any 
problem before messing with the update)..

What should the clever sysadmin do?

Thanks in advance,
Aldo

Try:

# standard asterisk command-line. No verbosity

   strace -eopen asterisk -U asterisk -c

See which module was the one last loaded.




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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

2014-01-29 Thread Ron Wheeler
I am surprised about the network. It should go before the disk if you 
have a lot of short transactions. There is a high percentage of overhead 
on streams of short messages.

Make sure that you check at each point where messages are passing.

Have you done any mathematical modeling of the disk and network traffic?

Try changing your RAM to see if raising it (or lowering it if that is 
easier) affects the problem.


What is your CPU utilization like at 80 calls?

How many open files do you have at 80 calls? is this near the limit?

Can you adjust the quality of the recordings to reduce the bits stored 
for each second of audio?

What happens when you do this?

Ron

On 29/01/2014 7:34 AM, Amit wrote:

Thanks Ron.
I will try to get these readings. About RAM disk, I will study on how 
to create RAM disk and conduct this test again.

There is no bottleneck on network.

After 80 calls, I see call drops, delay in responding, time out, 
re-transmission of SIP messages. If load is reduced, it settles again 
to normal.


*Thanks  Regards,*
Amit Patkar



On 1/28/2014 12:32 AM, Ron Wheeler wrote:
Can you get a reading of the total number of I/Os during your test? 
Peak IOPS?
That might tell you very quickly about the storage pattern that 
Asterisk uses.


Can you configure a RAM drive to see if disk is really the 
bottleneck. May need to add some more RAM memory to your configuration.


What is your network capacity? Usually one can write faster than the 
network can deliver - just to make sure that you are chasing the 
right bottleneck.


What happens at 80 calls to tell you that you have run out of IOPS?

Sorry for more questions than answers.

Ron



On 25/01/2014 12:26 AM, Amit wrote:

Thanks for response.
How do I derive the requirement? I need to size IO system to record multiple 
calls concurrently.
I ran test with following configuration
Quad Core Xeon with 4GB RAM
250GB SATA disk (No RAID)
Linux (CentOS 5.9)
Asterisk 1.8.20

I failed to record more than 80 calls.

If I run test with simple IVR, I achieved 400+ calls with same server.
So write seem to be an issue.
Is there any way to tune / optimize / configure for better write performance?

I am not sure if I need to post this query on developers list? Please guide...

Regards
Amit Patkar

Message: 1
Date: Fri, 24 Jan 2014 11:46:39 -0400
From: Mikeispbuil...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IOPS required by Asterisk for Call
Recording
Message-ID:52e28adf.8020...@gmail.com
Content-Type: text/plain; charset=iso-8859-1

On 14-01-24 11:16 AM, Amit wrote:

If I assume that Asterisk will write data on disk every second for
each call, I will need disk array to support minimum of 500 IOPS.
Where as if Asterisk push data every 2 seconds, I can deal with array
supporting 250 IOPS.
But if I assume that Asterisk will write data on disk for every RTP
packet received, as and when received, I will need disk IO system with
approx 25000 IOPS assuming 20 ms RTP packet.

You're assuming that asterisk will perform an fsync() after each write.
If asterisk writes without an fsync after each write, then the OS will
schedule writes intelligently based on RAM/disk IO available rather than
scheduling each one as a separate write.

Looking at the code for ast_writestream() there doesn't appear to be an
fsync() type call after each write, but someone more familiar with the
internals of Asterisk would be better able to verify that.


--
Ron Wheeler
President
Artifact Software Inc
email:rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102








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skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

2014-01-27 Thread Ron Wheeler
Can you get a reading of the total number of I/Os during your test? Peak 
IOPS?
That might tell you very quickly about the storage pattern that Asterisk 
uses.


Can you configure a RAM drive to see if disk is really the bottleneck. 
May need to add some more RAM memory to your configuration.


What is your network capacity? Usually one can write faster than the 
network can deliver - just to make sure that you are chasing the right 
bottleneck.


What happens at 80 calls to tell you that you have run out of IOPS?

Sorry for more questions than answers.

Ron



On 25/01/2014 12:26 AM, Amit wrote:

Thanks for response.
How do I derive the requirement? I need to size IO system to record multiple 
calls concurrently.
I ran test with following configuration
Quad Core Xeon with 4GB RAM
250GB SATA disk (No RAID)
Linux (CentOS 5.9)
Asterisk 1.8.20

I failed to record more than 80 calls.

If I run test with simple IVR, I achieved 400+ calls with same server.
So write seem to be an issue.
Is there any way to tune / optimize / configure for better write performance?

I am not sure if I need to post this query on developers list? Please guide...

Regards
Amit Patkar

Message: 1
Date: Fri, 24 Jan 2014 11:46:39 -0400
From: Mikeispbuil...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IOPS required by Asterisk for Call
Recording
Message-ID:52e28adf.8020...@gmail.com
Content-Type: text/plain; charset=iso-8859-1

On 14-01-24 11:16 AM, Amit wrote:

If I assume that Asterisk will write data on disk every second for
each call, I will need disk array to support minimum of 500 IOPS.
Where as if Asterisk push data every 2 seconds, I can deal with array
supporting 250 IOPS.
But if I assume that Asterisk will write data on disk for every RTP
packet received, as and when received, I will need disk IO system with
approx 25000 IOPS assuming 20 ms RTP packet.

You're assuming that asterisk will perform an fsync() after each write.
If asterisk writes without an fsync after each write, then the OS will
schedule writes intelligently based on RAM/disk IO available rather than
scheduling each one as a separate write.

Looking at the code for ast_writestream() there doesn't appear to be an
fsync() type call after each write, but someone more familiar with the
internals of Asterisk would be better able to verify that.


--
Ron Wheeler
President
Artifact Software Inc
email:rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] stopping unwanted attempts

2014-01-19 Thread Ron Wheeler

fail2ban is so easy to set up, there is no reason not to set it up.

The geography problems are not so bad unless you have phones all over 
the world or people travelling with softphones to countries that you 
want to block.


It does not block incoming calls only people who want to mimic your own 
legitimate phones.



Ron

On 19/01/2014 9:40 AM, Steve Murphy wrote:




On Sat, Jan 18, 2014 at 3:59 PM, Steve Edwards 
asterisk@sedwards.com mailto:asterisk@sedwards.com wrote:


On Sat, 18 Jan 2014, Jerry Geis wrote:

I see MANY of these in my log files:

[Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from
'202 sip:202@X:5060' failed for '37.8.12.147:26832
http://37.8.12.147:26832' - Wrong password

What is the correct way to block these idiots so they
don't even get this far.


Use iptables to allow packets from your legitimate users, block
everybody else.

If you are dealing with a mobile user base or an extensive
geographic area, at least block the countries where you do not
expect traffic -- North Korea, China, xxxistan, etc.

Drop these at the front door (90% of the problem) and use fail2ban
to pick off the rest.


I see a problem here; firstly that it is no longer so simple to determine
the IP ranges of countries. Things have been fractured quite a bit; you
might have to hire out a service to determine true geographic origination.
Even then, if your service is a little behind, you might occasionally
feel the displeasure of users unable to talk to your servers. How will you
handle this, with a white-list? How much effort will you end up committing
to keeping your whitelist up to date?

Nextly, the well-financed operations running such probes need not use
machines in their native countries. There are plenty of US-based
machines that can be ( and are ) compromised.


In other words, don't forget the fail2ban part!

Here's another idea! How about changing your port from 5060 to something
different, maybe 7067 or some other number that is not popularly being 
used?

You'll provision your phones to use this port, and the scanners will not
find you. Seems a much simpler solution... but there are some drawbacks...
can anyone think of them? And will these drawbacks matter to you? And, 
given
this solution, will the odds that a scanner might find your machine be 
so low,

that it is not worth using something like fail2ban to override them? Food
for thought!

murf

--

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Cody, WY 82414
?  murf at parsetree dot com
? 307-899-5535







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Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

2013-11-22 Thread Ron Wheeler
If you have no analog lines, Amazon/Rackspace/...  will probably beat 
your local ISP on bandwidth to your SIP/IAX carrier.


If your users are not in the same building as your in-house hosted 
Asterisk, Amazon might have a lot better connectivity with your users.


You certainly have a lot more flexibility in adding power to your setup 
at an Amazon.


I guess that one can decide what are the critical points that need to be 
tested (call volume, call quality, user connectivity) and devise a test 
setup.


Ron


On 22/11/2013 1:18 PM, Todd R. wrote:
I would have said the same thing a while back but, I can't ignore the 
fact that there have been what seems to be many Virtualization 
success stories.


The idea that Asterisk just likes to be on it's own dedicated hardware 
has always caused me to prefer dedicated hardware.


But, is the possibility of a single piece of hardware failing better 
than something that will likely never just flat out die?


I know there are high availability solutions out there and it's not 
that I don't have backups and disaster recovery plans in place.


I just want to make things far better regarding redundancy, recovery 
and scalability and virtualization is hard to beat when you start 
talking about these things.


There are definitely people/companies using virtualized Asterisk 
solutions successfully, so I feel like it can be done.


Asterisk has come a long way since I first starting messing with 
Asterisk and so has Asterisk itself.


So, I am trying to determine what is bad, what to look out for in 
terms of virtualizing. If it's still as bad of an idea as it was say 5 
years ago, then I need to understand why and if there is a work around.


At this point, the benefits of virtualizing my Asterisk boxes are too 
many to count. So, if I can't find any concrete reasons to NOT do this 
beyond That's a bad idea then I am going to give it a go. If I do, I 
am looking for any advice good or bad from those that have gone down 
this road successfully or with miserable failure.


My opinion all along has been Asterisk + Virtualization + Real Live 
Production Use = BAD IDEA!


Now, I am trying to figure out if that's just the opinion of an old 
man (sort of old) who just doesn't want to accept that virtualization 
if a better way (in terms of Asterisk).


So, I am hoping for people to tell me why Amazon AWS specifically is a 
good or bad idea with as much detail as possible.


Thanks!

 To: tjrl...@live.com; asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Amazon, Asterisk and reliability 
beyond a hobby system?

 Date: Fri, 22 Nov 2013 13:04:44 -0500
 From: cov...@ccs.covici.com

 I would thinktwice about Amazon -- and virtual in general is not a good
 idea for this sort of thing. I have seen messages about bad results
 with amazon specifically.

 Todd R. tjrl...@live.com wrote:

  Just checking one more time to see if anyone has an opinion on 
this. I am primarily interested in using a cloud type setup such as 
Amazon AWS for the redundancy, easy backup and recovery options. It's 
not about price but the idea that it will be very hard for a single 
piece of hardware to ruin my day.

 
  From: tjrl...@live.com
  To: asterisk-users@lists.digium.com
  Date: Mon, 18 Nov 2013 18:33:38 -0600
  Subject: [asterisk-users] Amazon, Asterisk and reliability beyond 
a hobby system?

 
 
 
 
  Took me a while but I have finally embraced cloud computing and 
all the benefits.
  The only thing I have yet to feel comfortable about putting in the 
cloud is real live Asterisk boxes to be used in production. I know 
it's being done because as far as I know Twilio is using Amazon for 
their Asterisk boxes.
  I have read all the fun articles on building hobby type systems 
and that's all great.
  What I really need to hear is from those that have deployed 
Asterisk in Amazon or Digital Ocean and how many simultaneous calls 
they are pushing through it and what the call quality and reliability 
has been.
  Right now I am still using dedicated hardware but I could become 
much more redundant and scale much faster using Amazon or Digital Ocean.
  Thanks in advance for any information from those that have already 
been down this road...

 
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Re: [asterisk-users] Asterisk 1.8.24 : illegal instruction

2013-11-20 Thread Ron Wheeler

Is it possible that in your build you mixed 32 bit and 64 bit libraries?

Ron
On 20/11/2013 8:06 AM, Jonas Kellens wrote:

Hello,

I have installed asterisk 1.8.24 (from source) but I can not start up 
Asterisk :



[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]# /sbin/service asterisk status
asterisk dead but subsys locked
[root@sip32 admin]# /sbin/service asterisk restart
Stopping safe_asterisk:[ OK  ]
Shutting down asterisk: [FAILED]
Starting asterisk:
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]#
[root@sip32 admin]# /usr/sbin/asterisk -c
Illegal instruction


Why can I not start Asterisk ?


I also notice the following in /var/log/messages :

[root@sip32 admin-voipcenter]# tail -f /var/log/messages
Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode 
ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000]
Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode 
ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000]
Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode 
ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode 
ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode 
ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode 
ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode 
ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode 
ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode 
ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000]
Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode 
ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000]
Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode 
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]
Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode 
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]






Kind regards,
Jonas.





--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Asterisk 1.8.24 : illegal instruction

2013-11-20 Thread Ron Wheeler
I am not sure that this is the cause of your problem but I think that 
the message that you are getting can be caused by that.


You might want to check the build logs to be sure that you do not have a 
32 bit library installed.


32 bit libraries will work on 64 bit Linux but not when mixed with 64 
bit applications.


Ron

On 20/11/2013 8:15 AM, Jonas Kellens wrote:

Hello,

how can I mix libraries ?

I have installed prerequisites from yum and asterisk from source (make 
 make install).


My kernel :

[root@sip32 asterisk-1.8.24.0]# uname -a
Linux sip32.domain.tld 2.6.32-358.23.2.el6.x86_64 #1 SMP Wed Oct 16 
18:37:12 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux




Jonas.


On 20-11-13 14:11, Ron Wheeler wrote:

Is it possible that in your build you mixed 32 bit and 64 bit libraries?

Ron
On 20/11/2013 8:06 AM, Jonas Kellens wrote:

Hello,

I have installed asterisk 1.8.24 (from source) but I can not start 
up Asterisk :



[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]# /sbin/service asterisk status
asterisk dead but subsys locked
[root@sip32 admin]# /sbin/service asterisk restart
Stopping safe_asterisk: [  OK  ]
Shutting down asterisk: [FAILED]
Starting asterisk:
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]#
[root@sip32 admin]# /usr/sbin/asterisk -c
Illegal instruction


Why can I not start Asterisk ?


I also notice the following in /var/log/messages :

[root@sip32 admin-voipcenter]# tail -f /var/log/messages
Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode 
ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000]
Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode 
ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000]
Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode 
ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode 
ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode 
ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode 
ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode 
ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode 
ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000]
Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode 
ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000]
Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode 
ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000]
Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode 
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]
Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode 
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]






Kind regards,
Jonas.





--
Ron Wheeler
President
Artifact Software Inc
email:rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102







--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-29 Thread Ron Wheeler

On 28/10/2013 4:12 PM, Mark Wiater wrote:


On 10/28/2013 3:59 PM, Ron Wheeler said:

I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the 
Asterisk - No analogue.

I don't have any problems with IAX, but I hear some do.

I have now switched to SIP and will check the quality in the morning.


We have a very lightly loaded 60 Mbs cable link to the Internet that 
tests pretty close to that most of the time.
Bandwidth is less important than the overall quality of the internet 
link, latency and jitter. Either way, there is no QoS on the internet, 
all bets are off.


The codec can matter too. What are you using?

G711




I have not found any good tools to track down the causes of poor 
voice quality.

In my case, I have good incoming quality and terrible quality going out.
Oh, is your cable connection assymetric? Upload smaller than download? 
If so, that correlates to terrible audio, right?

Just ran a test 50 Mbps  download 10Mbps upload. Should be enough I hope.


That is, I can hear people perfectly well but they complain that my 
voice drops out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls 
me, I call them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so 
poorly on the same environment.


Googling Asterisk poor audio quality return several hundred 
thousand references
I'd not shoot asterisk yet. I'd focus on the internet connection and 
it's components (cable modem) first.


Good idea. I am sure that you are right but what to test and how are not 
clear.
I use asterisk all over the place. Mostly connected to PRI's and 
Carrier provided SIP trunks, with internet SIP trunks as backup. I get 
complaints on the Internet based SIP trunks sometimes, never on other 
other two.


I'd ask most of these questions of the OP too. Overall telephony 
design matters.







--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-28 Thread Ron Wheeler

I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the 
Asterisk - No analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that 
tests pretty close to that most of the time.


I have not found any good tools to track down the causes of poor voice 
quality.

In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my 
voice drops out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls me, I 
call them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so poorly 
on the same environment.


Googling Asterisk poor audio quality return several hundred thousand 
references


Ron
On 28/10/2013 2:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone 
service that is being provided.  The choppy phone calls, and drop outs 
are detrimental to our sales force.


I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine

More than enough bandwidth

Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use

Hire a consultant

Ditch the system and buy a pre-packaged system - RingCentral or
some such.

There are no local asterisk professionals who can help, and we are a 
little leery of opening up our system to outside consultants.


Anyone else face the above, and finally abandoned Asterisk for a 
commercial system?


We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the 
conference rooms.


Suggestions welcome.

Best

Eddie
--
Eddie H. Mikell
Senior Systems Engineer
RKG

Office: 434.970.1010 x 124
Email:emik...@rimmkaufman.com mailto:emik...@rimmkaufman.com

http://www.rimmkaufman.com
http://twitter.com/rimmkaufman 
http://www.linkedin.com/company/85385 
http://plus.google.com/104980442218952272663/posts 
http://www.facebook.com/rimmkaufman http://www.RKGblog.com








--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] What linux distro most popular for Asterisk

2013-10-17 Thread Ron Wheeler

Hard to give an answer but for us it is
Centos - 1, others - 0

Ron

On 17/10/2013 6:16 AM, emilianovazq...@gmail.com wrote:

Most tutorials over internet are based on Centos and Ubuntu.

Centos is the base distro of FreePBX, Elastix and Trixbox and always have a lot 
of users.

I use ubuntu.

Best regards.

Emiliano


Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/)

-Original Message-
From: binary dreamer binary.vor...@gmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 17 Oct 2013 11:53:51
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] What linux distro most popular for Asterisk




--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] RTP port ranges

2013-09-18 Thread Ron Wheeler

I only use 100 ports as well but we have a very low call volume.
I thought that I saw that you need to allocate 2 ports for every 
simultaneous call that you need to support.
The ports are free (no charge) and are UDP not TCP so you do not lose 
any TCP ports.


I am not sure what a hacker could do if they attacked these ports.

Ron

On 18/09/2013 2:29 PM, Ira wrote:

Re: [asterisk-users] RTP port ranges Hello Thorsten,

Tuesday, September 17, 2013, 1:05:15 AM, you wrote:


Where is it stated that you MUST use 1-2 ???

Someone else please ?



Well, I don't use that range. This is that part of my rtp.conf

rtpstart=16000
rtpend=16100

I knew I didn't need the default 25000 ports, in fact 100 is probably 
more than 10 times what I'll ever need.


Been working for for 5 years with those numbers. I decided when I 
first did this that if I used non standard ports I might be less 
susceptible to hacking. Probably not accurate, but I did it anyway.


-- Ira


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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] LUA

2013-07-18 Thread Ron Wheeler

  
  
On 18/07/2013 10:00 AM, jon pounder
  wrote:


  
  On 07/18/2013 09:56 AM, jacob.e.mi...@l-3com.com
wrote:
  
  





  I am attempting to setup my server to use
Lua for the dialplan (extentions.lua), but I am unable to
get the asterisk configure script to find the installation
of Lua on my box. I have downloaded the Lua sources from
the www.lua.org
site, and I have installed via the make linux install
command. I can execute lua scripts via the command line,
but asterisk configure script is unable to find the
installation of Lua.
  
  I am on a closed network, so no access to
the internet so I am not able to just install Lua using yum.

  
  
  you're kidding right ? Why not just plug in the box somewhere
  else, do your install and move it back ?
  

If you can not do that, you can make a yum repo on your isolated
computer. 
You will need to download the RPMs and put them in your repo. 
Once you have done that, yum will be happy to use it.
yum is set up to have a number of repos configured and a local one
is just fine.


Ron


  

  
  
  OS CentOS 6.4
  Asterisk version 1.8.13.0  11.4
  
  $ find / -name *lua*
  /usr/local/include/lua.h
  /usr/local/include/lua.hpp
  /usr/local/include/lualib.h
  /usr/local/include/luaconf.h
  /usr/local/lib/lua
  /usr/local/lib/liblua.a
  /usr/local/bin/luac
  /usr/local/bin/lua
  /usr/lib64/liblua-5.1.so
  /usr/bin/luac
  /usr/bin/lua
  
  
  
  Jacob Miles
  Software Engineer
  jacob.e.mi...@l-3com.com
  903.457.4422
  
  
  




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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
  

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Re: [asterisk-users] LUA

2013-07-18 Thread Ron Wheeler

  
  
I suspected that the restriction might
  be policy rather than technical.
  Is there anything that guides the loading of software via USB or
  DVDs on isolated machines or is my suggestion about a local yum
  repo, a workable solution?
  
  Ron
  
  On 18/07/2013 10:04 AM, Kevin Larsen wrote:


  
  
  
  
  
  From:   
jon pounder
j...@inline.net
  
  To:   
asterisk-users@lists.digium.com,
  
  
  Date:   
07/18/2013 09:00 AM
  
  Subject:  
 Re: [asterisk-users]
LUA
  
  Sent by:  
 asterisk-users-boun...@lists.digium.com
  
  
  
  
  
  On 07/18/2013 09:56 AM, jacob.e.mi...@l-3com.com
wrote:
  
  I am attempting to setup my server
to use
Lua for the dialplan (extentions.lua), but I am unable to get
the asterisk
configure script to find the installation of Lua on my box. I
have
downloaded the Lua sources from the www.lua.org
site, and I have installed via the make linux install command.
I
can execute lua scripts via the command line, but asterisk
configure script
is unable to find the installation of Lua.
  
  
  
  I am on a closed network, so no
access
to the internet so I am not able to just install Lua using yum.
  
  you're kidding right ? Why not just plug in the box somewhere
  else, do
  your install and move it back ?




OS CentOS 6.4

Asterisk version 1.8.13.0 
  11.4



$ find / -name *lua*

/usr/local/include/lua.h

/usr/local/include/lua.hpp

/usr/local/include/lualib.h

/usr/local/include/luaconf.h

/usr/local/lib/lua

/usr/local/lib/liblua.a

/usr/local/bin/luac

/usr/local/bin/lua

/usr/lib64/liblua-5.1.so

/usr/bin/luac

/usr/bin/lua







Jacob Miles

Software Engineer

jacob.e.mi...@l-3com.com

903.457.4422







While a valid question, I have
  worked
  with clients on a closed military base where temporarily
  moving a box that
  has been secured back to an unsecured network would get you
  thrown off
  base and most likely result in criminal charges being filed.
  Not saying
  that is what Jacob is up against, but there are reasons that
  once a box
  is put somewhere you can't just move it back.


  
  
  
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Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x

2013-06-03 Thread Ron Wheeler

It looks like your database configuration is missing in Asterisk.

It is making up information about the connection using defaault values  
as if it did not find any database configuration.


Ron

On 03/06/2013 10:49 AM, Olivier CALVANO wrote:

Hi

i have installed a new Asterisk server on Fedora. My first server use 
Asterisk 1.6.x with a MySQL CDR and

realtime.

I have a small problems, when i configure on the new server, the same 
information in MySQL, we have a error:


[Jun  3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: 
Failed to connect database server SSI on myhost.myserver.com 
http://myhost.myserver.com (err 2003). Check debug for more info.
[Jun  3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail 
not found in database.  This table should exist if you're using realtime.
[Jun  3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql 
database SSI on myhost.myserver.com http://myhost.myserver.com.
[Jun  3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql 
database SSI on myhost.myserver.com http://myhost.myserver.com.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database user found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database password found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database host found, using localhost via socket.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database name found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database port found, using 3306 as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No 
database socket found (and unable to detect a suitable path).


The exacly same config work on 1.6.x

and from the new server, the database access is Ok:

[root@voip-2 log]# !mys
mysql -h myhost.myserver.com http://myhost.myserver.com -u Asterisk 
-p SSI

Enter password:
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 5185
Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL)

Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights 
reserved.


Oracle is a registered trademark of Oracle Corporation and/or its
affiliates. Other names may be trademarks of their respective
owners.

Type 'help;' or '\h' for help. Type '\c' to clear the current input 
statement.


mysql select * from VoiceMail;
+--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+
| uniqueid | customer_id | context  | mailbox| password | 
fullname | email | pager | tz  | attach | 
saycid | dialout | callback | review | operator | envelope | 
sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | 
forcename | forcegreetings | hidefromdir | stamp   |

+--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+
..


anyone know the problems ?

thanks
olivier




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Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x

2013-06-03 Thread Ron Wheeler

Fix this.

[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database user found, using 'asterisk' as default.
[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database password found, using 'asterisk' as default.
[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database host found, using localhost via socket.
[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database name found, using 'asterisk' as default.
[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database port found, using 3306 as default.
[Jun  3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No 
database socket found (and unable to detect a suitable path).


Asterisk is telling you that you have not configured ANY database.

It is not worrying about what tables are in it because you have not even 
defined the database itself.
There is NO database at all so worrying about versions is not Asterisk's 
big problem..


The rest of the messages after that are a bit screwy because the 
routines producing the error are not aware that there is no database at 
all so they just complain about the piece that they know about.



Ron


On 03/06/2013 12:19 PM, Olivier CALVANO wrote:

No other idea ?




2013/6/3 Olivier CALVANO o.calv...@gmail.com 
mailto:o.calv...@gmail.com


Hi

i have installed a new Asterisk server on Fedora. My first server
use Asterisk 1.6.x with a MySQL CDR and
realtime.

I have a small problems, when i configure on the new server, the
same information in MySQL, we have a error:

[Jun  3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime:
Failed to connect database server SSI on myhost.myserver.com
http://myhost.myserver.com (err 2003). Check debug for more info.
[Jun  3 16:27:59] WARNING[3140] res_config_mysql.c: Table
VoiceMail not found in database.  This table should exist if
you're using realtime.
[Jun  3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to
mysql database SSI on myhost.myserver.com
http://myhost.myserver.com.
[Jun  3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to
mysql database SSI on myhost.myserver.com
http://myhost.myserver.com.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database user found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database password found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database host found, using localhost via socket.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database name found, using 'asterisk' as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database port found, using 3306 as default.
[Jun  3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL
RealTime: No database socket found (and unable to detect a
suitable path).

The exacly same config work on 1.6.x

and from the new server, the database access is Ok:

[root@voip-2 log]# !mys
mysql -h myhost.myserver.com http://myhost.myserver.com -u
Asterisk -p SSI
Enter password:
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 5185
Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition
(GPL)

Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights
reserved.

Oracle is a registered trademark of Oracle Corporation and/or its
affiliates. Other names may be trademarks of their respective
owners.

Type 'help;' or '\h' for help. Type '\c' to clear the current
input statement.

mysql select * from VoiceMail;

+--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+
| uniqueid | customer_id | context  | mailbox| password |
fullname | email | pager | tz  | attach |
saycid | dialout | callback | review | operator | envelope |
sayduration | saydurationm | sendvoicemail | delete | nextaftercmd
| forcename | forcegreetings | hidefromdir | stamp

Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x

2013-06-03 Thread Ron Wheeler

Do you have this problem in your conf file?

http://forums.digium.com/viewtopic.php?p=63736

The parser won't accept an ; (semicolon) for remarks! So he found at 
the first the old remarks and tried to access my database with the false 
data.





Ron

On 03/06/2013 3:18 PM, Olivier CALVANO wrote:
on this server we don't have mysql.socket because he don't have mysql 
server


we want access to a mysql based on a other server




2013/6/3 Bakko asannu...@gmail.com mailto:asannu...@gmail.com

Hello,

are you sure MySQL socket is in /tmp directory?

dbsock = /tmp/mysql.sock

Regards

El 03/06/2013 12:16, Olivier CALVANO escribió:

Thanks for your help Ron,

Do you know where is the confirguration ?

Because i have put into res_config_mysql.conf:

[general]
dbhost = myhost.mydomain.net http://myhost.mydomain.net
dbname = MyDB
dbuser = MyUser
dbpass = MyPassword
dbport = 3306
dbsock = /tmp/mysql.sock
dbcharset = latin1
requirements = warn



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Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x

2013-06-03 Thread Ron Wheeler

Well, at least you are making progress.
What is the error in the debug log?

Ron


On 03/06/2013 8:03 PM, Olivier CALVANO wrote:
grrr no in asterisk -d i have no error, but when i start normaly 
asterisk i have :


[Jun  4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: 
Failed to connect database server xxx on xxx.xxx.net 
http://xxx.xxx.net (err 2003). Check debug for more info.
[Jun  4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: 
Failed to connect database server xxx on xxx.xxx.net 
http://xxx.xxx.net (err 2003). Check debug for more info.



what is the command in asterisk for i see the SQL query ?




2013/6/4 Olivier CALVANO o.calv...@gmail.com 
mailto:o.calv...@gmail.com


oh ron thanks for your help :

We have deleted all commented line, only put the configuration and
now that's work !




2013/6/3 Ron Wheeler rwhee...@artifact-software.com
mailto:rwhee...@artifact-software.com

Do you have this problem in your conf file?

http://forums.digium.com/viewtopic.php?p=63736

The parser won't accept an ; (semicolon) for remarks! So he
found at the first the old remarks and tried to access my
database with the false data.




Ron


On 03/06/2013 3:18 PM, Olivier CALVANO wrote:

on this server we don't have mysql.socket because he don't
have mysql server

we want access to a mysql based on a other server




2013/6/3 Bakko asannu...@gmail.com mailto:asannu...@gmail.com

Hello,

are you sure MySQL socket is in /tmp directory?

dbsock = /tmp/mysql.sock

Regards

El 03/06/2013 12:16, Olivier CALVANO escribió:

Thanks for your help Ron,

Do you know where is the confirguration ?

Because i have put into res_config_mysql.conf:

[general]
dbhost = myhost.mydomain.net http://myhost.mydomain.net
dbname = MyDB
dbuser = MyUser
dbpass = MyPassword
dbport = 3306
dbsock = /tmp/mysql.sock
dbcharset = latin1
requirements = warn



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mailto:rwhee...@artifact-software.com
skype: ronaldmwheeler
phone:866-970-2435, ext 102  tel:866-970-2435%2C%20ext%20102


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Re: [asterisk-users] Asterisk - Soundcard - Recording?

2013-05-28 Thread Ron Wheeler

What are you trying to accomplish?

What is the USB 'sound card' attached to?

Your description is too cryptic for someone to propose a solution.

Ron

On 28/05/2013 12:45 PM, Tim Nelson wrote:

Greetings-

I've got a curious project that I could use some input on. I'd like to use 
Asterisk to record some audio channels via USB 'soundcard'. When audio passes 
through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and 
write it to a wav file. I'm perfectly competent with the dialplan portion of 
the recording, but I don't know about the following:

-How does Asterisk know a new audio stream/source is beginning/ending?
-Can I have more than one CONSOLE device
-Is ALSA or OSS the preferred audio system with Asterisk these days (Asterisk 
11 presumably)

Any thoughts? Or, do you have any alternative ideas that would work better than 
using Asterisk for this?

Thanks!

--Tim

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Re: [asterisk-users] Asterisk - Soundcard - Recording?

2013-05-28 Thread Ron Wheeler

On 28/05/2013 1:23 PM, Tim Nelson wrote:

- Original Message -

What are you trying to accomplish?

What is the USB 'sound card' attached to?

Your description is too cryptic for someone to propose a solution.


The target use is to record mic level audio from various devices (could be an 
omnidirectional room mike, phone handset, etc).

--Tim

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Re: [asterisk-users] Asterisk - Soundcard - Recording?

2013-05-28 Thread Ron Wheeler

Sorry for the blank message. Fingers pressed send while brain was disenaged.

Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording

Ron


On 28/05/2013 1:23 PM, Tim Nelson wrote:

- Original Message -

What are you trying to accomplish?

What is the USB 'sound card' attached to?

Your description is too cryptic for someone to propose a solution.


The target use is to record mic level audio from various devices (could be an 
omnidirectional room mike, phone handset, etc).

--Tim

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Re: [asterisk-users] Asterisk - Soundcard - Recording?

2013-05-28 Thread Ron Wheeler
You are still being a bit evasive but should I understand that you want 
to run a headless machine with open microphones that records what ever 
it hears?

What do you want to do with each sound bite?
How long does the silence have to be before you close the recording and 
dispose of it (save, e-mail, upload, whatever).


Sounds like a security monitoring package (minus the video) should do 
the job?


A little Googleing shows up these.
http://oreka.sourceforge.net/about/
http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html

What else do you want it to do?

Ron

On 28/05/2013 2:23 PM, Tim Nelson wrote:

- Original Message -

Sorry for the blank message. Fingers pressed send while brain was
disenaged.

Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording


It would absolutely be a better solution. However, the recording is to be 
automated on a small system with no GUI, only console/SSH access. As such, 
running a full featured audio recording/mixing application in realtime (with 
user control) is not an option. :/

--Tim




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Re: [asterisk-users] Auto dialer scripts and software

2013-05-23 Thread Ron Wheeler
One of the tricks used in Canada was to call the other party's 
supporters and pretend that you are from their favorite party and piss 
them off.
You can also give them misleading information such as a phony change in 
voting location.


It appears to work since the guys who did this won the election.

If you hire a company outside your country to do this, you can make it 
hard to detect and impossible to prosecute.


Ron

On 23/05/2013 3:40 PM, cjwstudios wrote:
As long as you're dialing a screened registered voter list and don't 
call .gov or .edu, you're fine.



On Wed, May 22, 2013 at 5:54 AM, Don Kelly d...@donkelly.biz 
mailto:d...@donkelly.biz wrote:


Calls on behalf of political candidates are generally legal--even
to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.

--Don




-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Chris Bagnall
Sent: Wednesday, May 22, 2013 6:48 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Auto dialer scripts and software

On 22/5/13 10:54 am, A J Stiles wrote:
 You do know that sort of thing is against the law -- or at least
 requires a permit from the authorities -- in most civilised
countries,
right?

And it's worth adding that even if it is legal in your country, you're
almost guaranteed to offend/annoy your target audience. Recorded calls
always do.

Kind regards,

Chris
--
This email is made from 100% recycled electrons

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Re: [asterisk-users] looking for a way to do appointment reminders

2013-04-26 Thread Ron Wheeler

Good comment.
Another feature suggestion
You might to ask the person to press 1 to confirm or 2 to leave a 
message if the appointment is not going to be kept or 0 to reach the 
receptionist to reschedule the appointment.


Ron

On 26/04/2013 7:06 AM, Chris Bagnall wrote:

On 26/4/13 10:38 am, jg wrote:

they are currently calling patients. I think these calls apply only to a
certain fraction of the patients, who are difficult to contact by other
methods.


I suspect there will be different requirements depending on how 
'helpful' to patients you wish to be. At the very simplest end of the 
scale, you could simply call the patient's number and remind them of 
their appointment on dd hhmm, then disconnect.


However, the OP probably wants something a little more sophisticated 
than that. At the very least, you would want some method of handling 
shared numbers (e.g. a shared dwelling with a single phone), so you 
didn't inadvertently advertise a patient's appointment to someone else 
who answered the phone. So you would at the very minimum want a simple 
IVR that says We are trying to reach Mr. Joe Bloggs. If this is he, 
press 1 now, otherwise please hang up.


Going beyond that, you might want your reception staff, when booking 
appointments, to ask the patient when they would like their reminder 
call - the day before, an hour before, etc. etc. (and if the day 
before, would they prefer it in the morning, afternoon, or evening).


As others have said, the OP might be best advised to request (paid) 
assistance with the project on the [asterisk-biz] list.


Kind regards,

Chris



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Re: [asterisk-users] cdr report

2013-04-23 Thread Ron Wheeler

On 23/04/2013 11:09 AM, A J Stiles wrote:

On Tuesday 23 April 2013, aristidis tsitras wrote:

Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my calls
for the office. the problem is the management of the CDRs. i can see the
master.csv file, but it is not very friendly for the secretary of this
office to manage the calls.
is there a way to have a nice way to see the CDRs?Since the machine is
very small on CPU, it has to be as low on CPU/RAM consumption as possible.
any ideas?

CSV files can be opened with any spreadsheet software  (such as OpenOffice.org
calc or Numbers).

Alternatively, you can have the CDR using a database.  This can be on another
server.  Note if you are using MySQL, you will have to enable this yourself;
this is because not all of Asterisk is covered by the GPL, and the MySQL CDR
code ends up unredistributable.  (But it works as well as anything).  Then
write a Web app on the database server to display wanted CDR entries.

What about a script to convert the CSV to HTML and ftp the html file to 
a web server where it can be accessed as a browser page?



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Re: [asterisk-users] cdr report

2013-04-23 Thread Ron Wheeler

On 23/04/2013 11:42 AM, aristidis tsitras wrote:

On 04/23/2013 06:23 PM, Ron Wheeler wrote:

On 23/04/2013 11:09 AM, A J Stiles wrote:

On Tuesday 23 April 2013, aristidis tsitras wrote:

Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my 
calls
for the office. the problem is the management of the CDRs. i can 
see the

master.csv file, but it is not very friendly for the secretary of this
office to manage the calls.
is there a way to have a nice way to see the CDRs?Since the machine is
very small on CPU, it has to be as low on CPU/RAM consumption as 
possible.

any ideas?
CSV files can be opened with any spreadsheet software  (such as 
OpenOffice.org

calc or Numbers).

Alternatively, you can have the CDR using a database.  This can be 
on another
server.  Note if you are using MySQL, you will have to enable this 
yourself;
this is because not all of Asterisk is covered by the GPL, and the 
MySQL CDR
code ends up unredistributable.  (But it works as well as 
anything).  Then

write a Web app on the database server to display wanted CDR entries.

What about a script to convert the CSV to HTML and ftp the html file 
to a web server where it can be accessed as a browser page?





is it possible to have the script to convert to html? i will send it 
to a folder
echo htmlbodytable input.html; while read INPUT ; do echo 
trtd${INPUT//,//tdtd}/td/tr input.html ; done  
input.csv ; echo /table/body/html input.html


This Linux shell script will take your log after being copied to 
input.csv and make it into an ugly web page called input.html.


You can add more HTML to add a css style sheet and some column headings 
to make it look nice.


You probably want to fix the file names to reflect the date.


Ron



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Re: [asterisk-users] Logging SIP connection status for review

2013-04-10 Thread Ron Wheeler

http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.


Ron

On 10/04/2013 2:02 PM, Steve Edwards wrote:

On Wed, 10 Apr 2013, Carlos Alvarez wrote:

Is anyone using something to log SIP results (connected/not, latency) 
that they really like?  We do some logging using simple scripts 
writing the results of sip show peers to a text file if customers 
report issues, but it would be nice to have a tool that logs all the 
time and lets us do some better reporting.  For example, graphs of 
latency in a time range, or a list of unreachable phones within a 
range, etc.


dumpcap can capture all of the SIP (and RTP) packets into a series of 
files without a huge performance hit.


A cron job can pbzip2 the files and delete if over x days old.



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[asterisk-users] Reporting Utility

2013-01-15 Thread Ron Wheeler
When CDR reporting was raised a few days ago, it prompted me to add a 
section on how ADTransform could be used to address the problem raised 
about consolidating CDR information from various divisional PBXs and 
producing consolidated reports.


I wrote a short Use Case article.
http://www.artifact-software.com/?page_id=1666

I also thought about the problem of getting the configuration files into 
a readable format such as phone lists that could be distributed and 
added a second Use Case.
I am only maintaining our internal PBX so reporting configurations is 
not a big issue but I would imagine that for someone maintaining many 
clients or many corporate PBXs, having a batch tool that can collect the 
data files, produce nice looking reports and automatically upload them 
to a portal might be helpful.


I would be grateful for any comments about content, format or my skill 
as an artist.:-)


Ron

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Re: [asterisk-users] Which tool to edit custom reports from CDR and queues logs ?

2013-01-11 Thread Ron Wheeler

We are just delivering version 2 of our ADTransform data connector.
It would allow your to read in your CDR files, manipulate them, validate 
them and put out JasperReports based on the data.


It has a plug-in based workflow engine so that file transfers, 
input,transformation, validation, logging, reporting and automated 
report/file delivery steps can be added as required.


Out of the box it supports CSV and Excel files for input and output and 
other connectors such as JBDC, OBDC, webservices, extraction using 
custom APIs can be added  as plug-ins.

It supports JasperReports out of the box.
It supports FTP for getting and sending data and e-mail for sending 
reports, logs and data.


I am guessing that you might want to suck in all of the CDRs, apply some 
mappings to add location or user specific information for reporting and 
then run a series of JasperReports to provide summary and detail 
reports, possibly extract data for billing/chargeback and the deliver 
the data and reports through e-mail to the appropriate recipients.


It is a pure Java application that is OS agnostic.
We are looking at Raspberry support as part of version 3.

The original motivation for the product was the LMS market where data 
needs to be integrated from Payroll and HRIS to be feed into the LMS and 
certification and other training history data needs to be extracted to 
go to work scheduling or HRIS systems.


As a small Asterisk user, I think that I can understand where your 
requirement is coming from.


http://www.artifact-software.com/?page_id=929 is the website link if you 
want more info.
A short brochure is available. If anyone wants one, please contact me 
off-list.


Ron


On 11/01/2013 5:22 AM, Olivier wrote:

Hi,

I would like to edit reports showing how fast operator and users 
answer incoming calls.

Users are spread over 6 locations, each with its own asterisk instance.
Operator is on main site.
Users have casual extension but operator logs as queue agent.

I've read or/and tried Star2Billing's CDR-Stats and A2Billing, 
Asternic Call center Stats,.
I'm wondering if using a BI tool such as Jasper Reports would be 
preferable.


Which tool would you suggest to build custom reports ?

Regards


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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Ron Wheeler
://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Ron Wheeler

That does not solve any asterisk issue that I have.


On 10/01/2013 1:32 PM, Carlos Alvarez wrote:
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com 
mailto:r...@coopvr.com wrote:


It really didn't bother me as much as reading all the posts but
that's just me...now back to Asterisk issues :)


Sorry to add another, but for me, the main point is that this activity 
speaks to the character, ethics, and trustworthiness of the company 
doing it.  We all have spam filters.  I just also add the company to 
my do not buy/do not recommend list.



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TelEvolve
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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Ron Wheeler
We are about to announce the availability of a product that was written 
to solve data integration problems in Learning Management implementations.
It might help people who need to merge Asterisk CDR data with trunk 
provider's data and feed a CRM or billing system.


I would like to think that I could broadcast it here.

If it solves a problem, talk to me, otherwise I hope that people here 
will treat it with the equanimity that I show to people who talk about 
topics that do not apply to me.


Is there a lower limit of members that a question or statement must 
apply to before it is not spam?


I belong to another forum that has a lot of very strict rules about 
participation of vendors.
They are going to lose a lot of sources of good information and 
expertise if they enforce these rules.
They are almost antagonistic to vendors but as a vendor, I deal with the 
issues every day whereas the end-users are often hitting the problem for 
the first and only time in their lives.
As a vendor in other forums, I try to balance commercial interest with 
technical help.
I will usually let people know that I sell a certain product so that 
they understand that I have a commercial interest.

Then I give them my opinion or technical information.
I think that this is fair and lets the other members make a judgment 
about my statement with at least some knowledge about my bias and, 
perhaps, the limit of applicability of my suggestion.


OTOH, People  who try to push weight loss solutions here, deserve 
whatever gets thrown at them.


Ron

On 10/01/2013 5:32 PM, C. Savinovich wrote:


Isn't this precisely the raison d'être for [asterisk-biz]?

Oh my goodness!, the asteriz-biz?  nooo, they will kill you if you try 
to post anything offering your services!...  that list ceased to 
provide any value and died a long time ago precisely because its 
members ran each other away from it. A while back, I wrote a nice 
click-to-call  service and I dared put a post indicating that I was 
offering it for a fee, and in no time they called me spammer.  There 
is really no incentive to reward someone else's achievements, unless 
you tell them that you are given them your code for free, then they 
want it (totally contradicting the meaning of the word business).



Christian Savinovich
*/VoIP  Telephony Consultant/*
646-982-3572


 Original Message 
Subject: Re: [asterisk-users] DIDForSale spam
From: Chris Bagnall aster...@lists.minotaur.cc
mailto:aster...@lists.minotaur.cc
Date: Thu, January 10, 2013 5:17 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com

On 10 Jan 2013, at 22:09, C. Savinovich
c.savinov...@itntelecom.com mailto:c.savinov...@itntelecom.com
wrote:

 Unfortunately, there is a fine line between being a forum where
people can exchange ideas, and being a forum where people can find
asterisk consultants, and both don't seem to co-exist well together.

Isn't this precisely the raison d'être for [asterisk-biz]?


Kind regards,

Chris
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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Ron Wheeler

On 10/01/2013 5:34 PM, chris wrote:

There is a big difference between publicly posting offering services
to the list and harvesting all the email addresses and them contacting
everyone privately
We have to understand that we are going to be approached by people who 
think that we need their services.
Whether they get our e-mail here or from our web-sites or  from our 
business cards, we will get approached.


We are free to buy or not to buy. Read or ignore.

In a forum like this, that is almost the only price of free advice.


Ron



On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich
c.savinov...@itntelecom.com wrote:

Isn't this precisely the raison d'être for [asterisk-biz]?

Oh my goodness!, the asteriz-biz?  nooo, they will kill you if you try to
post anything offering your services!...  that list ceased to provide any
value and died a long time ago precisely because its members ran each other
away from it. A while back, I wrote a nice click-to-call  service and I
dared put a post indicating that I was offering it for a fee, and in no time
they called me spammer.  There is really no incentive to reward someone
else's achievements, unless you tell them that you are given them your code
for free, then they want it (totally contradicting the meaning of the word
business).


Christian Savinovich
VoIP  Telephony Consultant
646-982-3572



 Original Message 
Subject: Re: [asterisk-users] DIDForSale spam
From: Chris Bagnall aster...@lists.minotaur.cc
Date: Thu, January 10, 2013 5:17 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com

On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:


Unfortunately, there is a fine line between being a forum where people can
exchange ideas, and being a forum where people can find asterisk
consultants, and both don't seem to co-exist well together.

Isn't this precisely the raison d'être for [asterisk-biz]?


Kind regards,

Chris
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Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-03 Thread Ron Wheeler

On 03/01/2013 11:04 AM, Jeff LaCoursiere wrote:

On 01/03/2013 09:56 AM, Carlos Alvarez wrote:
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com 
mailto:myo...@acsacc.com wrote:



Where I am at is that they want us to use an SBC.  One engineer
asked about Cisco Call Manager.  I told them that basically if I
can accomplish the same thing with a Linux box (routing box and
sip proxy box) without having to spend money on SBCs or expensive
Cisco gear, that is the route we would like to go.  We are
looking at the possibility of handling 140 concurrent calls...
that is what they are designing on their end as well.

So, I am asking the community for any input.  I have read on here
and seen on IRC that some in the community are successfully using
Asterisk with Verizon SIP.  Verizon was going to check and see if
they have any notes about that and those particular setups.  Can
anyone help share any information or tidbits on how they were
able to sucessfully work with Verizon?


It may be too late for this, but in working with another RBOC who 
didn't want to deal with Asterisk, I just asked what they do support, 
and modified the headers sent by Asterisk to claim that it was one of 
the devices on that list.  Done.





ROFL!!  Well done.

j


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Re: [asterisk-users] Top Posting

2013-01-02 Thread Ron Wheeler

On 02/01/2013 1:11 PM, Carlos Alvarez wrote:
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com 
mailto:i...@extrasensory.com wrote:



And I started communicating with a 2400 baud modem so trimming was
a necessity and a requirement of friendship.


Bah, spoiled kids.  Mine was a 110 baud acoustic.

I think the Will Asterisk run on a Rasberry Pi thread the
perfect example of why this list is dying.


The number of questions posted here that are easily answered with a 
search or which are far too basic and open (how do I make Asterisk 
work) is very high these days, and that does kill a list.  A lot of us 
are interested in helping people who help themselves, and solving 
complex problems.  I've seen many tech lists die off when people stop 
trying to help themselves and ask intelligent questions.


As to top-posting, another example of when I think it's generally 
acceptable is people using tablets.  I have found no way on either my 
iOS or Android tablets to quickly/easily post in the traditional 
manner.  If I'm faced with spending a few minutes carefully trimming a 
useful reply or just not posting it at all, I'm likely to choose the 
latter if I'm on a list that says absolutely never top post.


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TelEvolve
602-889-3003



If you are answering one of my questions, please feel free to top post, 
bottom post or post in the middle.

I would rather have an answer than nothing - no matter how nicely formatted.

Part of the problem is the way that Asterisk is delivered.
The configuration files are way too complex and handle a lot of obscure 
situations rather than being minimal working configurations.


I am not sure that all of the defaults actually make sense - I just had 
to go in and turn on tos in SIP. The default is none which is not what 
the docs that I found, recommend.

SIP login comes with defaults that are not recommended for security reasons.

The documentation is hard to use.
At the same time, there is an expectation in the public that a competent 
system administrator can install an Asterisk PBX.


This being said, given the number of Asterisk installations being 
installed each day by first-time administrators, the traffic here seems 
pretty reasonable both in volume and in level of difficulty.


Ron

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[asterisk-users] Asterisk as answering machine

2013-01-02 Thread Ron Wheeler

I have connected a PSTN line to a Digium FXO card.
There is also an ordinary analogue phone attached to the same line.

The Asterisk answers the line on the first ring.

I would like it to wait for a few seconds so that someone can answer the 
PSTN line with an analogue phone.
This would allow a person to directly pick up the line if they wanted to 
or if not, let it go to the Asterisk where it would be dispatched 
through the normal process.


Currently, as soon as the analogue phone rings, the Asterisk PBX has 
already answered the call and starts the You have reached. Dial 
 and tries to dispatch the call.


This makes it hard to carry on a conversation.

Ron

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Re: [asterisk-users] Asterisk as answering machine

2013-01-02 Thread Ron Wheeler

On 02/01/2013 3:33 PM, Danny Nicholas wrote:

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, January 02, 2013 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk as answering machine

I have connected a PSTN line to a Digium FXO card.
There is also an ordinary analogue phone attached to the same line.

The Asterisk answers the line on the first ring.

I would like it to wait for a few seconds so that someone can answer the
PSTN line with an analogue phone.
This would allow a person to directly pick up the line if they wanted to or
if not, let it go to the Asterisk where it would be dispatched through the
normal process.

Currently, as soon as the analogue phone rings, the Asterisk PBX has already
answered the call and starts the You have reached. Dial  and tries
to dispatch the call.

This makes it hard to carry on a conversation.

Ron

In your dialplan, put Wait(10) in front of Answer().  This will give the
human 4 rings to pick up before Asterisk does.





I feel pretty silly. It worked. I saw this in a few Google responses but 
thought that I had already tested this.


Now I have just dumped myself back to newbee status.

Thanks
Ron

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Re: [asterisk-users] Top Posting

2012-12-30 Thread Ron Wheeler
I participate in a lot of lists and top posting is now the norm since 
people want to see quickly if the message is worth reading.
If the poster wants to intersperse comments in the following text, they 
announce that at the top so readers know to look further down.
Modern e-mail programs make it easy to figure out the history if you 
were not following the discussion closely.


Ron
On 29/12/2012 10:02 PM, Logan Bibby wrote:


I suppose I'm one of the few people that remember the content of 
threads by subject and easily catch up...


I'm also on my phone 99% of the time time and the way Gmail lays out 
emails makes top-posting beneficial to me.


On Dec 29, 2012 8:57 PM, Richard Kenner ken...@gnat.com 
mailto:ken...@gnat.com wrote:


 I realize the benefits of bottom-posting, especially when posting
 inline. But top-posting keeps things in reverse chronological order
 so any reader could catch up quickly on any missed messages in the
 chain. A new reader scrolls to the bottom and reads up.

What's there to catch up with if you don't first read what the
person
is replying to?  Do you think that everybody remembers every thread.
Of what value is it to see something like No, that didn't work.
*before*
a description of what it was that didn't work.

When people reply to an email, it's their responsibility, whether they
top-post or bottom-post to remove unnecessary old message and keep
just
what's necessary to understand the email.

One of the problems with top-posting is that it makes it easier to
forget
to do this.

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Re: [asterisk-users] Top Posting

2012-12-30 Thread Ron Wheeler



On 30/12/2012 11:13 AM, Patrick Lists wrote:

On 12/30/2012 04:26 PM, Ron Wheeler wrote:

I participate in a lot of lists and top posting is now the norm since
people want to see quickly if the message is worth reading.


Isn't it a bit of a stretch to extrapolate your experience with your 
lists to top posting being the norm? I am subscribed to several lists 
and bottom posting, proper trimming and commenting inline is the norm 
there.


Actually the norm is determined by the list rules. If the list rules 
say one must use bottom posting then one should use bottom posting. If 
someone does not like that then don't subscribe, find another source 
to ask a question (the forum, LUG, hire a consultant) or just bottom 
post.


Questions come before answers.
Answers come after questions.

-1 against changing rule #5.

Regards,
Patrick

Not really enough time in the day to keep track of different rules for 
all the forums.

I am more concerned about content than form.

As long as the questions get answered, I can figure out where it is but 
it is a PITA to scroll down through an e-mail to find out that there is 
nothing there worth reading.
I get over 100 e-mails per day that make it through my filters. I like 
to read the content as soon as it pops up rather than searching for the 
text.


Ron



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Re: [asterisk-users] Top Posting

2012-12-30 Thread Ron Wheeler

On 30/12/2012 3:54 PM, Benny Amorsen wrote:

Gergo Csibra csi...@gmail.com writes:


Complaining about top posting on a list where's no moderation,
no sanction if somebody top posting is pointless.

There is a sanction. People like me will score top posters lower and
soon not see their posts at all.

It is often a quick way to see if it is worth responding to someone. If
they top post, nothing of value is likely to come out of the
conversation.

So by all means, everybody who wants to, keep top posting.
Questions by their nature are all top posted and bottom posted so if you 
know any answers, your participation will be encouraged.

It is only long discussions that will miss your input.

Ron


/Benny


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Re: [asterisk-users] Top Posting

2012-12-30 Thread Ron Wheeler

I just wish that my biggest problem with Asterisk was top or bottom posting!

Ron

On 30/12/2012 9:45 PM, James Mortensen wrote:
Sorry for double posting, but I realized it was JIRA I spoke with 
Digium about, not Google Groups and the mailing list... However, I do 
think it's worth investigating or looking into alternatives that are 
more user friendly and that can make it easier to communicate with 
everyone on the list, whether a seasoned pro, top poster, or bottom 
poster.


James

On Sun, Dec 30, 2012 at 6:37 PM, James Mortensen 
james.morten...@voicecurve.com 
mailto:james.morten...@voicecurve.com wrote:


I have an idea! Instead of arguing over whether or not top posting
or bottom posting is the way to go, something that obviously no
one will /ever/ agree on, why not move to Google Groups instead
(or something similar to Google Groups).

When I post to Doubango's list, it's easy, there's no top or
bottom posting wars, it just works. In fact, in a thread, Google
Groups usually drops you right to the most recent message, so the
people who like top posting can still see the most recent message
while the bottom posters will still see the bottom posting format.

It's either this, or we can sit and watch intelligent people
continue to degrade one another and argue over something with no
agreement in site. :)

When I mentioned this before, someone from Digium said this will
never happen, and it's unfortunate.  Maybe they just like to see
people bicker and argue.

If there's a better alternative to Google Groups, or a way to set
preferences in the mailing list so that everyone is happy, maybe
that's something that could be done?

James
?



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Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Ron Wheeler

On 27/12/2012 3:14 PM, Carlos Alvarez wrote:
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.ca 
mailto:maill...@lightspeed.ca wrote:


This past holiday weekend has resulted in some real groaners when
it comes to bugs in our dialplan, making obvious the need for some
changes in our procedures.

First, our hours of operation for Christmas Eve, Christmas, Boxing
Day and New Year's Eve had changed with little to no notice. Okay,
fine, whatever, I fix.


Boxing day???  Seriously?  There's a holiday for people who beat each 
other up?  TIL.
That is the day you box up all the crap you got and exchange it for what 
you really wanted.
It is a religious holiday in the old British Commonwealth (probably 
Scottish in origin).


Ron



But anyway the best way to test time-based rules is on a VM that has a 
copy of your configs, and just change the time.  You can easily run a 
small VM to accomodate a copy of your main server on almost any computer.


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602-889-3003



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Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

2012-12-06 Thread Ron Wheeler

It seems like a safe thing to do.
You could also ask about the impact of making an existing column a  
primary key, in a MySQL forum.


Leandro's solution seems to be a good one as well and does guarantee 
uniqueness.




Ron

On 06/12/2012 12:25 PM, Leandro Dardini wrote:
Yes, go for it. However I have added another autoincrement column and 
created the primary key on it. On the other columns I need to search I 
have created just an index.


Leandro

2012/12/6 Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr

Hello,

I need to develop an application that will query (mostly reading)
an existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx
2.10 install on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.

The tools I'm planning to use require this table to include a
Primary Key.
Is it safe to Alter this table telling it to use UniqueID column
as a Primary Key ?

(Sure, I'll test this on a database copy but I'm not confident my
tests will cover everything)

Regards

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Re: [asterisk-users] Questions about extension.conf

2012-11-29 Thread Ron Wheeler

On 29/11/2012 11:47 AM, Salman Zafar wrote:

It is self explanatory, for example:

exten =  _X.,1, Noop(Let say we have allowed all numbers i.e. _X 
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and 
thats it)



Hope I have answered your question.

Not for me.
What part of those lines and comments discusses same?

What is the syntax for a same line? what does it mean to use same 
rather than exten?






On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long longst...@gmail.com 
mailto:longst...@gmail.com wrote:


Hello

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)})  ;
make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)
  ; filter out bad characters per the
README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org
http://freenum.org)}) ; perform our lookup with freenum.org
http://freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
  ; check if we set the FREENUMDOMAIN global variable in [global]

same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
   if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)

exten = fn-BUSY,1,Busy()

exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()

;##


According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;

Syntax for defining a context: keywords *exten*, *include*,
*ignorepat* and *switch*. same is not mentioned in this wiki.

There is a part of dial plan from sample extension.conf above. My
Question is  how same = key word works .

Thanks



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Re: [asterisk-users] Questions about extension.conf

2012-11-29 Thread Ron Wheeler

That is a good answer.
Thanks.
Any reason why it is not documented?

Ron

On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:


Shitian Long wrote 29.11.2012 18:40:

There is a part of dial plan from sample extension.conf above. My 
Question is  how same = key word works .

Thanks
  


same is used for complex templates, if you don't want to copy 
previous line or afraid you can make a typo.


exten = _1XXNXXX,1,Answer

same = n,HangUp

is the substitution for:

exten = _1XXNXXX,1,Answer

exten = _1XXNXXX,n,HangUp

Also,  it makes grepping the particular exten in a file a lot easier, 
and if you want to change some template for exten which has 50 lines, 
you don't have to edit all 50 of them.


--
With Best Regards
Mikhail Lischuk  mailto:mlisc...@itx.com.ua

  



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Re: [asterisk-users] Questions about extension.conf

2012-11-29 Thread Ron Wheeler

Excellent.
It appears that Getting Started has a lot more stuff in it than the 
documentation for 1.8.


Very helpful.

Ron

On 29/11/2012 12:31 PM, David M. Lee wrote:


On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:


That is a good answer.
Thanks.
Any reason why it is not documented?


It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities


Ron


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Re: [asterisk-users] SIP not answering on one trunk.

2012-11-28 Thread Ron Wheeler

I did some more testing.
The SIP phone does answer and it stops ringing. The incoming call 
(DAHDI) keeps broadcasting a ring tone to the caller's handset even 
though the SIP phone stops ringing when the call is picked up and shows 
that the call is connected.

Sorry for the confusion.

If the SIP phone does not answer, the call goes through the voicemail 
process and takes a message.


Does that help narrow it down?
It looks like I have done something to the Asterisk configuration that 
is preventing the bridging of the incoming call to the local extension.
It rings the local, the local picks up but the incoming caller is not 
connected and still hears the pbx ringing the SIP phone even though the 
SIP phone is no longer actually ringing.


Ron

On 28/11/2012 1:15 PM, Joshua Colp wrote:

Ron Wheeler wrote:

I have 2 analog trunks.

They answer the incoming call, do the welcome message, ask for the
extension, when a valid extension is entered it rings the right SIP
phone BUT
when the SIP phone is answered, the SIP phone keeps ringing and the call
is not connected.
If the phone is not answered it goes to voicemail correctly.


I would suggest you grab a sip set debug on trace for this issue to 
confirm that the signaling is fine. When the phone answers it should 
send a 200 OK to Asterisk and then we respond with an ACK. If that all 
looks correct, as it seems to from the log you have provided thus far, 
I think you may have a phone specific issue.


Cheers,




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Re: [asterisk-users] SIP password probe

2012-11-27 Thread Ron Wheeler

On 27/11/2012 12:58 PM, Christopher Harrington wrote:
It's an open source project. Pay a programmer or make the modification 
yourself and submit a patch.

You don't really want me coding!
I have solved the problem for me.

Just add it to the queue of enhancements for the next time someone is 
working on SIP.


Ron




On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler 
rwhee...@artifact-software.com 
mailto:rwhee...@artifact-software.com wrote:


I looking through my logs, I found that people where probing my
SIP accounts looking for passwords.
Asterisk was helping them out by processing hundreds of requests
per minute.
I did a bit of Googling and this seems to be a frequent knock
against Asterisk's security.

It would seem pretty simple to add a configuration setting to
sip.conf to delay the response to a bad account or password.

There is a half measure to confuse the probe by sending the same
error return for either error.
It appears that many people have complained that this should be
the default setting only changed if your are debugging a problem.

There is no reason for a working system to ever have bad passwords
so this is clearly an attack in almost every case.

A simple delay would solve the problem for most people who use
reasonable passwords.

I had to install fail2ban which is a PITA but thanks to someone's
clear recipe, I was able to get it working.

I hope that this can be worked into a release soon.

Ron

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Re: [asterisk-users] SIP password probe

2012-11-27 Thread Ron Wheeler

I had to install fail2ban and configure it to watch Asterisk.

Ron

On 27/11/2012 2:11 PM, Mitul Limbani wrote:


You might want to share the know how over here if its not a chan_sip 
patch.


Mitul

On Nov 28, 2012 12:28 AM, Ron Wheeler 
rwhee...@artifact-software.com 
mailto:rwhee...@artifact-software.com wrote:


On 27/11/2012 12:58 PM, Christopher Harrington wrote:

It's an open source project. Pay a programmer or make the
modification yourself and submit a patch.

You don't really want me coding!
I have solved the problem for me.

Just add it to the queue of enhancements for the next time someone
is working on SIP.

Ron




On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler
rwhee...@artifact-software.com
mailto:rwhee...@artifact-software.com wrote:

I looking through my logs, I found that people where probing
my SIP accounts looking for passwords.
Asterisk was helping them out by processing hundreds of
requests per minute.
I did a bit of Googling and this seems to be a frequent knock
against Asterisk's security.

It would seem pretty simple to add a configuration setting to
sip.conf to delay the response to a bad account or password.

There is a half measure to confuse the probe by sending the
same error return for either error.
It appears that many people have complained that this should
be the default setting only changed if your are debugging a
problem.

There is no reason for a working system to ever have bad
passwords so this is clearly an attack in almost every case.

A simple delay would solve the problem for most people who
use reasonable passwords.

I had to install fail2ban which is a PITA but thanks to
someone's clear recipe, I was able to get it working.

I hope that this can be worked into a release soon.

Ron

-- 
Ron Wheeler

President
Artifact Software Inc
email: rwhee...@artifact-software.com
mailto:rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102 tel:866-970-2435%2C%20ext%20102


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President
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email:rwhee...@artifact-software.com  
mailto:rwhee...@artifact-software.com
skype: ronaldmwheeler
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skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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[asterisk-users] SIP not answering on one trunk.

2012-11-27 Thread Ron Wheeler

I have 2 analog trunks.

They answer the incoming call, do the welcome message, ask for the 
extension, when a valid extension is entered it rings the right SIP 
phone BUT
when the SIP phone is answered, the SIP phone keeps ringing and the call 
is not connected.

If the phone is not answered it goes to voicemail correctly.


[DID_866nnn]  IAX that works
include = DID_866nnn_default
[DID_866nnn_default]
exten = 866n,1,Goto(voicemenu-artifact-en,s,1)

[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default]
exten = s,1,Goto(voicemenu-artifact-fr,s,1)

[DID_trunk_2]
include = DID_trunk_2_default
[DID_trunk_2_default]
exten = s,1,Goto(voicemenu-home,s,1)

[voicemenu-artifact-en]
;ArtifactEnglishFirst
include = default
include = conferences
exten = s,1,Answer
exten = s,n,Set(CALLERID(name)=Art-${CALLERID(name)})
exten = s,n,Wait(0.5)
exten = s,n,Background(record/HelloArtifactEnglish)
exten = s,n(menu),Background(record/DialExtensionEnglish)
exten = s,n,WaitExten(3)
exten = 0,1,Goto(inbound-reception,s,1)
exten = 9,1,Goto(changeLanguageFrArtifact,s,1)
exten = #,1,Directory(default,default,f)
exten = t,1,Goto(inbound-reception,s,1)
exten = i,1,Goto(voicemenu-artifact-en,s,menu)

My IAX trunks work.

Log of dialing in on Trunk2 - answering SIP 102 and waiting while it 
continued to ring.


[2012-11-27 14:43:52] VERBOSE[3589] sig_analog.c: -- Starting simple 
switch on 'DAHDI/2-1'
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing 
[s@DID_trunk_2:1] Goto(DAHDI/2-1, voicemenu-home,s,1) in new stack

[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Goto (voicemenu-home,s,1)
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing 
[s@voicemenu-home:1] Answer(DAHDI/2-1, ) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing 
[s@voicemenu-home:2] Set(DAHDI/2-1, CALLERID(name)=Home-ARTIFACT 
LOGICI) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing 
[s@voicemenu-home:3] Wait(DAHDI/2-1, 0.5) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing 
[s@voicemenu-home:4] BackGround(DAHDI/2-1, 
record/HelloAnnetteAndRon) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] file.c: -- DAHDI/2-1 Playing 
'record/HelloAnnetteAndRon.ulaw' (language 'en')

[2012-11-27 14:43:56] VERBOSE[3589] pbx.c:   == CDR updated on DAHDI/2-1
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing 
[102@voicemenu-home:1] Macro(DAHDI/2-1, stdexten,102,SIP/102) in new 
stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing 
[s@macro-stdexten:1] Set(DAHDI/2-1, __DYNAMIC_FEATURES=) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing 
[s@macro-stdexten:2] Set(DAHDI/2-1, ORIG_ARG1=102) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing 
[s@macro-stdexten:3] GotoIf(DAHDI/2-1, 0?6:4) in new stack

[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Goto (macro-stdexten,s,4)
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing 
[s@macro-stdexten:4] Dial(DAHDI/2-1, SIP/102,20,) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] netsock2.c:   == Using SIP RTP CoS 
mark 5

[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: -- Called SIP/102
[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: -- SIP/102-0002 
is ringing
[2012-11-27 14:44:01] VERBOSE[3589] app_dial.c: -- SIP/102-0002 
answered DAHDI/2-1

Rang for a whole minute and a half  until I hung up the DAHDI/2-1
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c: -- Executing 
[h@voicemenu-home:1] Hangup(DAHDI/2-1, ) in new stack
[2012-11-27 14:45:42] VERBOSE[3589] features.c:   == Spawn extension 
(voicemenu-home, h, 1) exited non-zero on 'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] app_macro.c:   == Spawn extension 
(macro-stdexten, s, 4) exited non-zero on 'DAHDI/2-1' in macro 'stdexten'
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c:   == Spawn extension 
(voicemenu-home, 102, 1) exited non-zero on 'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] sig_analog.c: -- Hanging up on 
'DAHDI/2-1'

[2012-11-27 14:45:42] VERBOSE[3589] chan_dahdi.c: -- Hungup 'DAHDI/2-1'
(END)

--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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[asterisk-users] SIP password probe

2012-11-24 Thread Ron Wheeler
I looking through my logs, I found that people where probing my SIP 
accounts looking for passwords.

Asterisk was helping them out by processing hundreds of requests per minute.
I did a bit of Googling and this seems to be a frequent knock against 
Asterisk's security.


It would seem pretty simple to add a configuration setting to sip.conf 
to delay the response to a bad account or password.


There is a half measure to confuse the probe by sending the same error 
return for either error.
It appears that many people have complained that this should be the 
default setting only changed if your are debugging a problem.


There is no reason for a working system to ever have bad passwords so 
this is clearly an attack in almost every case.


A simple delay would solve the problem for most people who use 
reasonable passwords.


I had to install fail2ban which is a PITA but thanks to someone's clear 
recipe, I was able to get it working.


I hope that this can be worked into a release soon.

Ron

--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] How to set SIP to auto answer in the dial plan .

2012-07-14 Thread Ron Bergin
upendra wrote:
 Hi,


 I am trying to write dial plan for sip to auto answer (auto attend) the
 incoming call to the sip phone.

 - If i call from sip1 to sip2 then sip2 should automatically answer the
 call and play some sound file.
 I am trying to do this but as new to the asterisk dial plan configuration
 ,
 so not able Todo this.
 help me if anyone already done this setup.



 Regards
 Upendra.
 --

Unless I'm misunderstanding your needs, wouldn't this do what you want?

exten = 1234,1,Answer
exten = 1234,n,Playback(soundfile)
exten = 1234,n,Dial(SIP/1234,60,m)  ; caller hears music on hold
 ; instead of ringtone

-- 
Ron Bergin



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Re: [asterisk-users] DAHDI DTMF problem?

2012-07-06 Thread Ron Bergin
Bill Dunn - VCI Internet Services wrote:
 I have an Asterisk server configured to run as voicemail with a T1 and
 SMDI.
 It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a
 few
 years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on
 Centos 5.8

 The problem I am having appears to be related to DTMF detection. When the
 test phone number is called (2704083000) Asterisk only receives a portion
 of
 the dialed number. It varies as to what numbers are detected. Sometimes it
 sees a single digit, sometimes 3 or 4 of the digits of the dialed number.

 When I compare this to the old server the debug below is similar but there
 isn't any mention of the sig_analog.c lines shown below.

 I am told the T1's on the old server and the new server are configured the
 same. I can make outgoing calls on the T1 from Asterisk.

 Can someone give me a clue as to what could be causing this?


 Bill Dunn


Try setting:
relaxdtmf=yes

We used to have that same problem on most of our servers.  Setting
relaxdtmf to yes solved the problem for us.

-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.




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Re: [asterisk-users] DAHDI DTMF problem?

2012-07-06 Thread Ron Bergin
Bill Dunn - VCI Internet Services wrote:
 Thanks Ron. I have had my chan_dahdi.conf file set as follows with the
 same
 result.

 [trunkgroups]
 [channels]
 switchtype=national
 usecallerid=yes
 callerid=asreceived
 cidsignalling=smdi
 echocancel=yes
 echocancelwhenbridged=yes
 relaxdtmf=yes
 rxgain=0.0
 txgain=0.0
 usesmdi=yes
 smdiport=/dev/ttyS0
 signalling = em_w
 immediate = no
 group = 1
 channel = 1-3



  Bill Dunn



 - Original Message -
 From: Ron Bergin
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Friday, July 06, 2012 12:34 PM
 Subject: Re: [asterisk-users] DAHDI DTMF problem?


 Try setting:
 relaxdtmf=yes

 We used to have that same problem on most of our servers.  Setting
 relaxdtmf to yes solved the problem for us.


Are you using SIP?  If so, relaxdtmf can also be set in sip.conf as well
as a dtmfmode setting that you can adjust.

What type of phones are you using?  In our case, part of this problem was
due our low end 2.4ghz cordless phones.

-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics
(408) 487-4600



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Re: [asterisk-users] SCCP Questions

2012-06-14 Thread Ron McCarthy
Hello,

Thanks you for the replies ill take a look at the driver you sent over. Im
going to run some test and see what happens, hopefully the driver in 1.8 is
soild and nothing needs to be messed with, but we will see :)

On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote:

 Greetings Ron-

 Just wanted to give you a heads up about an alternative SCCP channel
 driver available for Asterisk. Please see here:

 http://freecode.com/projects/chan-sccp-b

 I have no experience with it (nor SCCP in general) but just wanted to give
 you an option in the event the included SCCP driver does not give you
 satisfactory results.

 --Tim

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Re: [asterisk-users] SCCP Questions

2012-06-14 Thread Ron McCarthy
Is the chan-sccp-b project the same one that got put in SVN of 1.8 branch?
I have not been able to find anything definitive that says so, I really
need 1.8 branch so trying to see which is the best way to go.

Thanks

On Thu, Jun 14, 2012 at 9:34 AM, Ron McCarthy ronmc...@gmail.com wrote:

 Hello,

 Thanks you for the replies ill take a look at the driver you sent over. Im
 going to run some test and see what happens, hopefully the driver in 1.8 is
 soild and nothing needs to be messed with, but we will see :)


 On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote:

 Greetings Ron-

 Just wanted to give you a heads up about an alternative SCCP channel
 driver available for Asterisk. Please see here:

 http://freecode.com/projects/chan-sccp-b

 I have no experience with it (nor SCCP in general) but just wanted to
 give you an option in the event the included SCCP driver does not give you
 satisfactory results.

 --Tim

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[asterisk-users] SCCP Questions

2012-06-13 Thread Ron McCarthy
Hi List,

Has anyone been running SCCP with a larger number of phones? Im looking to
deploy like 75+ phones and I want to keep SCCP so I don't have to upgrade
them and for the SLA, some phones also have no SIP software for them so im
forced to keep SCCP. Does anyone have any experience with this? From what
ive read the SCCP support works and works well, im just worried about
trying to run this many phones and if im missing any sort of issues that
could come up.

Thanks!
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Re: [asterisk-users] Best practices to route calls according holidays

2012-05-19 Thread Ron Bergin
Olivier wrote:
 Hi,

 At the moment, I'm mostly using a Day/Night toggle button to let
 users deal with week-ends, holidays and opening hours.
 As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if
 better alternatives now exist.

 Is it possible, safe, reliable and easy to refer from Asterisk to a
 public calendar resource listing holidays, for a given country ?
 Should you instead refer to a private resource, to avoid depending on
 an externaly managed resource ? If you go this way, which tools would
 you recommend to build and update a private calendar ?

 Suggestions ?

 Regards

 --

The database approach that others have suggested sounds pretty good.  What
I did was to write a simple agi script that dispatches a subroutine based
on the holiday.  I hard coded the holidays in the script, but they could
just as easily be stored in a db.

Here's the key portion of the script.  (The formatting may get goofed up
in the email).

#!/usr/bin/perl

use strict;
use warnings;
use Asterisk::AGI;
use Date::Calendar;

$|++;

my ($min, $hr, $day, $mo, $yr, $dow) = (localtime)[1..6];
$mo++;
$yr += 1900;
my $today = sprintf(%d%02d%02d, $yr,$mo,$day);

my $holidays = {
New Year's Day = #Jan/1,
Easter = +0,
Memorial Day   = 5/Mon/May,
Independence Day   = #Jul/4,
Labor Day  = 1/Mon/Sep,
Thanksgiving   = 4/Thu/Nov,
Black Friday   = 4/Fri/Nov,
Christmas Eve  = #Dec/24,
Christmas Day  = #Dec/25,
Christmas Dayafter = #Dec/26,
New Year's Eve = #Dec/31
};

my %dispatch = (
New Year's Day = \new_years_day,
Easter = \easter,
Memorial Day   = \memorial_day,
Independence Day   = \july4,
Labor Day  = \labor_day,
Thanksgiving   = \thanksgiving,
Black Friday   = \black_friday,
Blackout Period= \blackout_hrs,
Christmas Eve  = \christmas_eve,
Christmas Day  = \christmas_day,
Christmas Dayafter = \christmas_dayafter,
New Year's Eve = \new_years_eve,
);

my $agi= Asterisk::AGI-new;
my $calendar   = Date::Calendar-new( $holidays );
$calendar-year( $yr );

foreach my $holiday ( keys %$holidays ) {
my @holiday = $calendar-search( $holiday );
my $holidaydate = sprintf(%d%02d%02d, $holiday[0]-year,
$holiday[0]-month,
$holiday[0]-day
  );
if ( $today == $holidaydate ) {
$dispatch{ $holiday }-($agi);
exit;
}
}

if ( in_blkout_period( $today ) ) {
$dispatch{Blackout Period}-( $agi, $dow, $hr );
exit;
}

##

sub playback {
my ($agi, $holiday, $hrs) = @_;

$agi-stream_file([
   'frys/thank_you_for_calling',
   frys/$holiday,
   frys/$hrs,
   'frys/enjoy',
   'frys/frys_goodbye'
  ]
);
}

sub new_years_day {
my $agi = shift;

$agi-exec('noop', Incoming call on New Year's Day);
if ($hr  10 or $hr = 19) {
playback($agi, 'new_years_day', '10to7');
$agi-hangup();
}
else {
$agi-exec('Goto', 'welcome');
}
}


-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.



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Re: [asterisk-users] Reliable SIP Trunk Provider

2012-03-15 Thread Ron Bergin
Jake Wicke wrote:
 I'm wondering if any other Asterisk users have a recommendation for a
 reliable SIP Trunk provider that supports Asterisk and offers decent
 support.

 I've worked with Coredial, Broadvox, and Broadvoice and have had some bad
 experiences with each of these providers.


I'm going to assume that you're in the US, since those 3 providers are all
based here.

I can highly recommend XO Communications. http://www.xo.com/

We currently have 35 SIP trunks with them and will be adding more.  Our
corp office is on a DS3 SIP trunk with 500 DID's and our stores are on a
T1 SIP trunk with 100 DID's.

They have several levels of support.  We use their upper level support
called SNA (I forget what it stands for), which gives us direct access to
their upper level engineers when needed.  Their front line support people
that I deal with are very good and may be VoIP engineers themselves.

Ron Bergin
Network Operations Administrator
Fry's Electronics Inc.


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Re: [asterisk-users] Compiling asterisk with mysql support

2012-03-08 Thread Ron Bergin
Jason Parker wrote:
 On 03/06/2012 12:31 PM, Ron Bergin wrote:

 Mathew,

 Each of those odbc modules are unavailable i.e., marked with XXX

 I even deleted the asterisk build directory and started over, but had
 the
 same results.

 What prereqs do I need besides these:

 mysql.i386  5.0.95-1.el5_7.1installed
 mysql-connector-odbc.i386   3.51.26r1127-1.el5  installed
 mysql-devel.i3865.0.95-1.el5_7.1installed
 mysql-server.i386   5.0.95-1.el5_7.1installed
 unixODBC.i386   2.2.11-7.1  installed
 unixODBC-devel.i386 2.2.11-7.1  installed


 libtool-ltdl-devel should be a dependency for unixODBC-devel in CentOS,
 but it
 is not.  You'll need to install that and re-run ./configure.

 --
 _


Jason,

Sorry for the delay in responding.  I just realized that my response,
which was shortly after yours, was stuck in limbo on my side.

Installing libtool-ltdl-devel fixed the odbc problem and asterisk is now
starting up fine.

Thanks
-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.


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[asterisk-users] Compiling asterisk with mysql support

2012-03-06 Thread Ron Bergin
I have a working asterisk test box that I'm rebuilding with mysql support
so that I can test cdr stats (http://www.cdr-stats.org/).

When I run 'make memuselect', I select res_config_mysql, app_mysql,
cdr_mysql, and app_saycountpl components.  The build/install process goes
fine i.e. no errors.  However, I'm getting a seg fault error when starting
asterisk.

# /usr/sbin/safe_asterisk: line 145: 27014 Segmentation fault  (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS}
 /dev/${TTY} 21  /dev/${TTY}

CentOS release 5.7
asterisk-1.8.9.2
dahdi-linux-complete-2.6.0
libpri-1.4.12

What am I missing?

-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.





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Re: [asterisk-users] Compiling asterisk with mysql support

2012-03-06 Thread Ron Bergin
Paul Belanger wrote:
 On 12-03-06 12:05 PM, Ron Bergin wrote:
  However, I'm getting a seg fault error when
 starting asterisk.

 # /usr/sbin/safe_asterisk: line 145: 27014 Segmentation fault  (core
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
 ${ASTARGS} /dev/${TTY} 21  /dev/${TTY}

 What am I missing?

 For what ever reason, asterisk is crashing.  You'll need to generate a
 backtrace[1].

 [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

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

Thanks Paul, I'll look at getting the backtrace a little later today.


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Re: [asterisk-users] Compiling asterisk with mysql support

2012-03-06 Thread Ron Bergin
Matthew  Jordan wrote:

 - Original Message -
 From: Ron Bergin r...@i.frys.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, March 6, 2012 11:05:35 AM
 Subject: [asterisk-users] Compiling asterisk with mysql support


 When I run 'make memuselect', I select res_config_mysql, app_mysql,
 cdr_mysql, and app_saycountpl components.  The build/install process
 goes
 fine i.e. no errors.  However, I'm getting a seg fault error when
 starting
 asterisk.

 Please also note that app_saycountpl, res_config_mysql, app_mysql, and
 cdr_mysql are all either extended support modules or deprecated:

 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States

 Note that if the segmentation fault is occurring in an extended support
 module, development support typically comes from the Asterisk community.
 In the case of deprecated modules, your mileage will vary considerably.

 You may want to consider using an ODBC connection instead of interfacing
 directly with the MySQL database, using the following:

 * app_mysql = func_odbc
 * cdr_mysql = cdr_adaptive_odbc
 * res_config_mysql = res_odbc
 * app_saycountpl = say.conf


 What am I missing?

 --
 Ron Bergin
 Network Operations Administrator
 Fry's Electronics, Inc.


 Matthew Jordan
 Digium, Inc. | Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --

Thanks for the info.  Can I assume that these (mysql) modules will be
removed soon or will they come alive again down the road?

-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.



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Re: [asterisk-users] Compiling asterisk with mysql support

2012-03-06 Thread Ron Bergin
Matthew  Jordan wrote:

 - Original Message -
 From: Ron Bergin r...@i.frys.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, March 6, 2012 11:05:35 AM
 Subject: [asterisk-users] Compiling asterisk with mysql support

 I have a working asterisk test box that I'm rebuilding with mysql
 support
 so that I can test cdr stats (http://www.cdr-stats.org/).

 When I run 'make memuselect', I select res_config_mysql, app_mysql,
 cdr_mysql, and app_saycountpl components.  The build/install process
 goes
 fine i.e. no errors.  However, I'm getting a seg fault error when
 starting
 asterisk.

 Please also note that app_saycountpl, res_config_mysql, app_mysql, and
 cdr_mysql are all either extended support modules or deprecated:

 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States

 Note that if the segmentation fault is occurring in an extended support
 module, development support typically comes from the Asterisk community.
 In the case of deprecated modules, your mileage will vary considerably.

 You may want to consider using an ODBC connection instead of interfacing
 directly with the MySQL database, using the following:

 * app_mysql = func_odbc
 * cdr_mysql = cdr_adaptive_odbc
 * res_config_mysql = res_odbc
 * app_saycountpl = say.conf

 --
 Ron Bergin
 Network Operations Administrator
 Fry's Electronics, Inc.


 Matthew Jordan
 Digium, Inc. | Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --

Mathew,

Each of those odbc modules are unavailable i.e., marked with XXX

I even deleted the asterisk build directory and started over, but had the
same results.

What prereqs do I need besides these:

mysql.i386  5.0.95-1.el5_7.1installed
mysql-connector-odbc.i386   3.51.26r1127-1.el5  installed
mysql-devel.i3865.0.95-1.el5_7.1installed
mysql-server.i386   5.0.95-1.el5_7.1installed
unixODBC.i386   2.2.11-7.1  installed
unixODBC-devel.i386 2.2.11-7.1  installed

-- 
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.



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