Re: [Asterisk-Users] MySQL and Asterisk

[Matthew Boehm]

Dan Journo wrote:
Is there a guide anywhere which runs through how to set up asterisk with 
mysql?
 
I've looked and almost all the document misses out relevant information.
 
Thanks
 
Dan Journo


What do you want to do with mysql? Did you read on the wiki? There is 
tons of info there.


-Matthew

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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Matthew Boehm]

   CPU0   CPU1
  0: 85 1703809954IO-APIC-edge  timer
  8:  0  0IO-APIC-edge  rtc
  9:  0  1   IO-APIC-level  acpi
 14:  0 31IO-APIC-edge  ide0
177:  0   17840313   IO-APIC-level  megaraid
185:  0 1817423967   IO-APIC-level  eth0
193:  0   40198530   IO-APIC-level  eth1
201:  0 3507106255   IO-APIC-level  wanpipe1, wanpipe2, wanpipe3,
wanpipe4
NMI:  0  0
LOC: 1633394197 1633394188
ERR:  0
MIS:  0

Any idea what "LOC" and "timer" are? I did "watch -n1 cat /proc/interrupts"
for about a minute. The largest movers were "timer", "LOC" and "eth0". The
others either never change or hardly changed.

-Matthew

> From: Sig Lange <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Date: Mon, 19 Sep 2005 15:34:26 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
> 
> It means you have a piece of hardware that is generating a lot of interupts.
> Try a few commands like this:
> 
> # vmstat 1
> # watch -n1 cat /proc/interrupts
> Go through lspci -vb and disable hardware that's not being used.
> 
> Watch the numbers as they increase. Also check for ERR and MIS
> 
> 
> On 9/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
>> 
>> 
>>> OK. Perhaps I was not clear. Please read my original post again:
>>> 
>>> I've been trying to diagnose why my server has a constant idle time
>> of 90%
>>> even when nothing is running.
>>> 
>>> I did not say "90% usage" I said "90% idle". Meaning I have a constant
>> CPU
>>> usage of 10%.
>>> 
>>> This 10% measurement comes from the "hardware interrupt" (hi) from
>> within
>>> "top":
>>> 
>>> Cpu(s): 3.8% us, 2.1% sy, 0.0% ni, 85.5% id, 0.2% wa, 8.0% hi, 0.4% si
>>> 
>>> 
>>> Even when all other percentages are at 0%, hi remains around 10%. How
>> can I
>>> figure out what is causing all these interrupts?
>> 
>> I don't have a clue other then to unload drivers, etc.
>> 
>> 
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> 
> 
> 
> -- 
> Sig Lange
> http://www.signuts.net/
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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Matthew Boehm]

OK. Perhaps I was not clear. Please read my original post again:

 I've been trying to diagnose why my server has a constant idle time of 90%
 even when nothing is running.

I did not say "90% usage" I said "90% idle". Meaning I have a constant CPU
usage of 10%.

This 10% measurement comes from the "hardware interrupt" (hi) from within
"top":

Cpu(s):  3.8% us,  2.1% sy,  0.0% ni, 85.5% id,  0.2% wa,  8.0% hi,  0.4% si


Even when all other percentages are at 0%, hi remains around 10%. How can I
figure out what is causing all these interrupts?

-Matthew


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Re: RE : [Asterisk-Users] Asterisk realtime beta

[Matthew Boehm]

That is not a limitation of the Asterisk RealTime Architecture. That is a
limitation of libiodbc.

-Matthew


> From: Olivier Taylor <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 19 Sep 2005 18:09:38 +0200
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: RE : [Asterisk-Users] Asterisk realtime beta
> 
> The limitation is that it doesn't work on freebsd, probably due to
> libiodbc...
> That's a limitation, isn't it?
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Matthew Boehm
> Envoyé : lundi 19 septembre 2005 16:42
> À : Asterisk Users
> Objet : Re: [Asterisk-Users] Asterisk realtime beta
> 
> 
> So, you admit that you can do what you want using RealTime Static, but you
> are just unwilling to do so. So, how is that a limitation if you 'can' do
> it?
> 
> -Matthew
> 
>> From: Urban <[EMAIL PROTECTED]>
>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Date: Mon, 19 Sep 2005 11:03:09 +0200
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Subject: Re: [Asterisk-Users] Asterisk realtime beta
>> 
>> Matthew Boehm wrote:
>> 
>>>> I currently not use it due to some limitations in * realtime .
>>>>
>>>> 
>>> 
>>>Such as?
>>> 
>>> -Matthew
>>> 
>>> 
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>>>  
>>> 
>> My current configuration uses a lot of include statements to split up
>> the context's such as security contexts included per extension (allow
>> national, internation calls etc). Since realtime does not have this
>> type of feature (if you not using static) I decided it was to much
>> work to redeisgn the dialplan at the moment.
>> 
>> /urban
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[Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Matthew Boehm]

I've been trying to diagnose why my server has a constant idle time of 90%
even when nothing is running.

After finally discovering what "hi" means in 'top' (it means hardware
interrupts) I find that this percentage always averages around 7-10%.

How can I find out what is causing this constant load of interrupts?

I have a Dell 1850 3.0Ghz with on board RAID and 2GB RAM.

Anyone else experiencing this?

-Matthew


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Re: [Asterisk-Users] Asterisk realtime beta

[Matthew Boehm]

So, you admit that you can do what you want using RealTime Static, but you
are just unwilling to do so. So, how is that a limitation if you 'can' do
it?

-Matthew

> From: Urban <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 19 Sep 2005 11:03:09 +0200
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] Asterisk realtime beta
> 
> Matthew Boehm wrote:
> 
>>> I currently not use it due to some limitations in * realtime .
>>>
>>> 
>> 
>>Such as?
>> 
>> -Matthew
>> 
>> 
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>>  
>> 
> My current configuration uses a lot of include statements to split up
> the context's such as security contexts included per extension (allow
> national, internation calls etc). Since realtime does not have this type
> of feature (if you not using static) I decided it was to much work to
> redeisgn the dialplan at the moment.
> 
> /urban
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Re: [Asterisk-Users] Asterisk realtime beta

[Matthew Boehm]

> I currently not use it due to some limitations in * realtime .

Such as?

-Matthew


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Re: [Asterisk-Users] Pass through of T.38

[Matthew Boehm]

Carlos Alperin wrote:

I don't get this?

Is included on 1.0.9 or is not? I know that a lot of people was trying it,
but just to be clear, is T.38 passthrough included on 1.0.9?

Thanks,

Carlos Alperin


No. We are not even sure at this point if T38 will make it into 1.2.

-Matthew

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Re: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb

[Matthew Boehm]

cytrex2*CLI> cdr mysql status
Connected to [EMAIL PROTECTED], port 3306 using table cdr for 3 days, 18
hours, 4 minutes, 3 seconds.
  Wrote 62028 records since last restart.

That is what you should see.

-Matthew

> From: alexandre zhang <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 12 Sep 2005 04:07:28 +0800 (CST)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb
> 
> thanks for ur help
>  
> Run the command "cdr mysql status"
> I got the following msg
> ' No such command 'cdr mysql' (type 'help' for help)'
>  
> But, I run ' show modules'
> cdr_addon_mysql.so  is in the list
>  
> Do u have an idea  about it ?
>  
> Thanks
> 
> 
> 
> Matthew Boehm <[EMAIL PROTECTED]> 写道：
> Run the command "cdr mysql status" from asterisk CLI. What does that say? If
> it says command not found then the module is not loaded.
> 
> -Matthew
> 
> 
>> From: alexandre zhang
>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST)
>> To: 
>> Subject: [Asterisk-Users] cdr_addon_mysql.so pb
>> 
>> hi 
>> 
>> I load cdr_addon_mysql.so without error
>> 
>> configuration of cdr_mysql.conf
>> 
>> [general]
>> dbhost = localhost
>> dbname = recharge
>> dbuser = root
>> dbpass = ast
>> dbport = 3306
>> dbsock = /var/lib/mysql/mysql.sock
>> 
>> 
>> But, I get nothing in the table of cdr of my database.
>> 
>> 
>> Somebody have an idea ? Thanks you for your help
>> 
>> best regards
>> 
>> 
>> 
>> -
>> DO YOU YAHOO!?
>> 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
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> 
> -
> DO YOU YAHOO!?
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Re: [Asterisk-Users] cdr_addon_mysql.so pb

[Matthew Boehm]

Run the command "cdr mysql status" from asterisk CLI. What does that say? If
it says command not found then the module is not loaded.

-Matthew


> From: alexandre zhang <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST)
> To: 
> Subject: [Asterisk-Users] cdr_addon_mysql.so pb
> 
> hi 
>  
> I load cdr_addon_mysql.so without error
>  
> configuration  of cdr_mysql.conf
>  
> [general]
> dbhost = localhost
> dbname = recharge
> dbuser = root
> dbpass = ast
> dbport =  3306
> dbsock = /var/lib/mysql/mysql.sock
> 
>  
> But, I get nothing in the table of cdr of my database.
>  
>  
> Somebody have an idea ? Thanks you for your help
>  
> best regards
>  
> 
> 
> -
> DO YOU YAHOO!?
>   雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
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Re: [Asterisk-Users] Pass through of T.38

[Matthew Boehm]

Roger Schreiter wrote:

Hi,

I found some contradicting infos about pass through of
T.38 data.


	I almost had to change my pants when I saw a CVS update this morning 
adding T38 frame recognition to asterisk. I kept looking for the code 
that complimented this but haven't seen it yet. And there was no bug 
reference so I can't help test.


-Matthew

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[Asterisk-Users] Polycom 501 Multiple Line Instances

[Matthew Boehm]

I tried following the Wiki page regarding the Polycom 501 and having the 
same extension appear on all 3 line buttons (just like my cisco) but I'm 
having no luck.


Has anyone else had success in doing this? Perhaps someone who has been 
successful can update the wiki?


Thanks,
Matthew

http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501

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Re: [Asterisk-Users] MAX PRI for single server

[Matthew Boehm]

[EMAIL PROTECTED] wrote:


Ah - so the difference between your setup and mine is that you are using
Sangoma (presumably) and I'm using Digium.  Looks like the Digium is
significantly more efficient then.


	It could also be that I'm using Net-SNMP to query my cpu usage and even 
when the machine is idle, SNMP reports about 20% CPU usage which is 
incorrect.


-Matthew

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Re: [Asterisk-Users] MAX PRI for single server

[Matthew Boehm]

Carlos Antunes wrote:

Have you seen this?
http://www.digium.com/index.php?menu=compatibility


Yes, but I'm not using a Digium card.

-Matthew

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Re: [Asterisk-Users] MAX PRI for single server

[Matthew Boehm]

Jason Becker wrote:

Hmm, looks like someone "in the know" needs to update the wiki:

http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P


Wow. Guess I'm not.

	I've got a 4 port PRI card in this brand new Dell 1850 3.0Ghz Xeon with 
2GB RAM and I run an average of 50-60% CPU usage with just 47 calls. All 
G711.


I guess a 64 bit chip is really that much better eh?

-Matthew

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Re: [Asterisk-Users] MAX PRI for single server

[Matthew Boehm]

Jason Becker wrote:

Sage advice, but out of curiousity what happened to Digium's T3 card 
(the DS3000P)?


	IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it 
will have no on-board EC and no on-board encoding so I can't imagine the 
machine you would need to process that many calls.


-Matthew

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Re: [Asterisk-Users] Re: MAX PRI for single server

[Matthew Boehm]

Wayne Gemmell wrote:

On Thursday 08 September 2005 16:26, Simone Cittadini wrote:


My boss is just asking me if it is possible to stuck 4* TE411P in a


Doesn't that equal 16 lines, not 480 lines? Or did I miss something?



Yes, you missed something:

4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines

That is assuming you have 1 D-chan per span.

-Matthew

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Re: [Asterisk-Users] Multiple Instances of Asterisk (no contexts)

[Matthew Boehm]

Geoff Karl wrote:

I know I have seen something on the mailing list describing how to run
more than one instance of Asterisk.  I can't find it anymore.

What are the things to look for when running more than one copy.

Yes, I know about contexts.

thanks,

Geoff


	This begs a repeated question: Why? The entire 'point' of contexts is 
so you don't have to run multiple instances of asterisk.


-Matthew

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Re: [Asterisk-Users] Insert Subject Here

[Matthew Boehm]

Flobi wrote:
I've been messing with it for a couple weeks with MySQL.  It seems 
pretty good to me though I have had a couple crashes.  I cane' say for 
sure that the crashes were directly related to RealTime though.  Also, 
I'm still using CVS HEAD 2005-09-06 which was right before the beta 
release, I think. 


Flobi, please use the subject line. Its there for a reason.

Secondly, if your system is crashing, how do you expect us to help debug 
the problem if you don't provide any info? Like backtrace's etc..


Read doc/README.backtrace for more info.

-Matthew

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Re: [Asterisk-Users] Extensions - Realtime

[Matthew Boehm]

The wiki doc's are correct. You are trying to combine two different methods
of pulling RealTime extensions and that is why it isn't working as you are
expecting.

Pick 1 method and all will be revealed. Both are very simple to do.

-Matthew

> From: Flobi <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Date: Wed, 7 Sep 2005 13:00:26 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] Extensions - Realtime
> 
> Nevermind, I figured out that the table is used way differently when
> doing static.  Here's my fixed table.  I'll try to explain this in the
> voip-info doc.
> 
>   id  cat_metric  var_metric  commented  filename  category  var_name  var_val
>   1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing)
> 
> 
> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote:
>> Okay, after noticing an error on this mysql statement after i switched to
>> odbc:
>> SELECT * FROM
>> pbx_realtime_extensions
>> WHERE filename='extensions.conf' and commented=0
>> ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id
>> 
>> I added those fields and reloaded...* immediately crashed.  I
>> restarted.  Now, I'm getting this:
>> *CLI> show dialplan
>> [ Context 'NoOp' created by 'pbx_config' ]
>> 
>> [ Context 'parkedcalls' created by 'res_features' ]
>>  '700' =>  1. Park()
>> [res_features]
>> 
>> -= 1 extensions (1 priorities) in 2 contexts. =-
>> 
>> 
>> out of this table:
>>  id  name  context  exten  priority  app  appdata  filename
>> commented  cat_metric  var_metric  category  var_name  var_val
>>  1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL
>> NULL NULL NULL
>> 
>> 
>> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote:
>>> Okay, this doesn't seem to be working.  I've gone and deleted my ael
>>> file also.  I do know my MySQL is set up cause I have my sip, iax and
>>> voicemail going through it too.
>>> 
>>> here's the line in extconfig.conf:
>>> [settings]
>>> extensions.conf => mysql,asterisk,pbx_realtime_extensions
>>> 
>>> 
>>> in pbx_realtime_extensions, my db table:
>>> id  name  context  exten  priority  app  appdata
>>> 1  default  default  _.  1  NoOp  Testing
>>> 
>>> 
>>> CLI> show dialplan
>>> [ Context 'parkedcalls' created by 'res_features' ]
>>>  '700' =>  1. Park()
>>> [res_features]
>>> 
>>> -= 1 extensions (1 priorities) in 1 contexts. =-
>>> 
>>> And when I try to call, I get:
>>> Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
>>> connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
>>> exist
>>> 
>>> Also, this message keeps popping up even when calls aren't going through:
>>> Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
>>> find extension context 'default'
>>> 
>>> 
>>> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote:
>>>> 
>>>> It states that the conf file overrides the static db info, but what about
>>>> the ael file?  Does that override also?
>>>> 
>>>> BTW, "RealTime Static"...talk about oxymoron :-)  Gotta love it!
>>>> 
>>>> Flobi
>>>> 
>>>> 
>>>> On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
>>>>>> CVS HEAD/Asterisk 1.2: Is there a way to have the entire
>>>>>> extensions.conffile coming from the realtime?
>>>>> 
>>>>>Yes. Go read the wiki on RealTime Static.
>>>>> 
>>>>> -Matthew
>>>>> 
>>>>> 
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>>>>> 
>>>>> Asterisk-Users mailing list
>>>>> Asterisk-Users@lists.digium.com
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>> 
>>>> 
>>>> 
>>>> 
>>>> 

Re: [Asterisk-Users] Extensions - Realtime

[Matthew Boehm]

I don't see your swich statement anywhere.

You must define a context [default] then add in the correct switch=>
statement.

-Matthew


> From: Flobi <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Date: Wed, 7 Sep 2005 12:18:26 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] Extensions - Realtime
> 
> Okay, this doesn't seem to be working.  I've gone and deleted my ael
> file also.  I do know my MySQL is set up cause I have my sip, iax and
> voicemail going through it too.
> 
> here's the line in extconfig.conf:
> [settings]
> extensions.conf => mysql,asterisk,pbx_realtime_extensions
> 
> 
> in pbx_realtime_extensions, my db table:
> id  name  context  exten  priority  app  appdata
> 1  default  default  _.  1  NoOp  Testing
> 
> 
> CLI> show dialplan
> [ Context 'parkedcalls' created by 'res_features' ]
>   '700' =>  1. Park()
> [res_features]
> 
> -= 1 extensions (1 priorities) in 1 contexts. =-
> 
> And when I try to call, I get:
> Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
> connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
> exist
> 
> Also, this message keeps popping up even when calls aren't going through:
> Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
> find extension context 'default'
> 
> 
> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote:
>> 
>> It states that the conf file overrides the static db info, but what about the
>> ael file?  Does that override also?
>>  
>> BTW, "RealTime Static"...talk about oxymoron :-)  Gotta love it!
>>  
>> Flobi
>>  
>> 
>> On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
>>>> CVS HEAD/Asterisk 1.2: Is there a way to have the entire
>>>> extensions.conffile coming from the realtime?
>>> 
>>>Yes. Go read the wiki on RealTime Static.
>>> 
>>> -Matthew
>>> 
>>> 
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>>> 
>> 
>> 
>> 
>> 
>> -- 
>> Automated Signature: This message is from Flobi of Flobi.com.
>> Visit my website if you like: http://www.flobi.com/
>> 
>> Please remember to tip your waitress and bartender.  They are doing their
>> best to serve you and your indignant, malcontent attitude.
>> -- 
> 
> 
> 
> -- 
> Automated Signature: This message is from Flobi of Flobi.com.
> Visit my website if you like: http://www.flobi.com/
> 
> Please remember to tip your waitress and bartender.  They are doing
> their best to serve you and your indignant, malcontent attitude.
> --
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Re: [Asterisk-Users] Extensions - Realtime

[Matthew Boehm]

> CVS HEAD/Asterisk 1.2: Is there a way to have the entire
> extensions.conffile coming from the realtime?

Yes. Go read the wiki on RealTime Static.

-Matthew


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Re: [Asterisk-Users] PHP and ASterisk Manager

[Matthew Boehm]

Anton Krall wrote:

Guys, is anybody using PHP sockets to connect to the Manager and send
command like "show voicemail users" for example or any other?

My question is, how to parse the return info in a way that can be shown back
to the user via web (discard all the manager responses not needed)?


Use preg_match() to match the lines you want the user to see on the website.

$socket = fsockopen("localhost","5038", $errno, $errstr, 30);

if(!$socket) {
print "No socket";
exit();
}

fputs($socket, "Action: Login\r\n");
fputs($socket, "Events: Off\r\n");
fputs($socket, "UserName: bleh\r\n");
fputs($socket, "Secret: bleh\r\n\r\n");

fputs($socket, "Action: Command\r\n");
fputs($socket, "Command: show channels\r\n\r\n");

fputs($socket, "Action: Logoff\r\n\r\n");

while(!feof($socket)) {
$buff = fgets($socket,1024);
if(preg_match("/SIP\/.*/", $buff)) {
print "I found a SIP call";
}
}

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Re: [Asterisk-Users] Cisco 7960 upgrades

[Matthew Boehm]

You cannot go from 5.3 -> 7.5. You must go from 5.3 -> 7.0 then to 7.5.

-Matthew

> From: Sascha Ferley <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 5 Sep 2005 13:19:40 -0600
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: [Asterisk-Users] Cisco 7960 upgrades
> 
> Hi, 
> 
>  
> 
> I got a problem of having to upgrade 35 Cisco 7960 phones from default
> firmware of 3.1 to 7.5.
> The problem I get is that when trying to upgrade I see on the tftplog that
> it can't seem to find the file (8 character issue).
> 
> So I renamed the files to suit what is supposed to be in them.
> 
>  
> 
> I am trying incremental upgrades from 3.1 -> 5.3 -> 7.5, with no luck. It
> goes to Upgrading Software and sits there endlessly redownloading the same
> file. 
> 
> It seems to stall going no-where .. Any one successfully upgraded the phones
> from default? What are any of the specifics. With the 5x series do I need
> the P003-05 in the OS79XX.TXT file or still the P0S3 ?
> 
>  
> 
> Anyone have any ideas as to what I should do? I can't seem to get the pre 5x
> versions of the software any more. Seems with my contract I can only
> downgrade to 5x series. All that shows on the Cisco CCO site.
> 
>  
> 
> Please let me know
> 
> Thanks
> 
>  
> 
> Sascha
> 
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Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration

[Matthew Boehm]

> MySQL RealTime Static seems to see the settings as it goes through and
> does the select.. but the it just kinda ignores them

Strange. Have you verified this behavior with ODBC RealTime? The code
that parses the results is virtually identical so I don't see this as a
mysql-rt specific issue.

-Matthew


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Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration

[Matthew Boehm]

How did you convert your voicemail.conf file into RT Static? Did you use the
perl script?

-Matthew


> From: Matt <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sun, 4 Sep 2005 20:37:34 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
> 
> I should add to this... I understand to make the table.. but when I
> make it.. asterisk selects it but seems to ignore things.   No where
> have I found documented what the var_category and such are... what
> numbers do I put in there?!?!
> 
> On 9/4/05, Matt <[EMAIL PROTECTED]> wrote:
>> Hi,
>> When using asterisk real-time with mysql voicemail integration...
>> where exactly do I put the options like the [PBX] tag, and how long
>> silence can be, etc?
>> 
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Re: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration

[Matthew Boehm]

You must store voicemail.conf using RealTime Static in order to use the
options you have mentioned from database.

-Matthew

> From: Matt <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sun, 4 Sep 2005 19:51:36 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration
> 
> Hi,
> When using asterisk real-time with mysql voicemail integration...
> where exactly do I put the options like the [PBX] tag, and how long
> silence can be, etc?
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Re: [Asterisk-Users] PRI Identity Crisis

[Matthew Boehm]

Damon Estep wrote:

Sounds like you are connected to a pbx instead of a carrier switch,
either asterisk or the pbx need to be set to network

Pri_cpe or pri_net in asterisk...


Oh, you are good.

-Matthew

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[Asterisk-Users] PRI Identity Crisis

[Matthew Boehm]

Just upgraded to most recent CVS and now I get this:

pri_dchannel: PRI Error: We think we're the CPE, but they think they're 
the CPE too.


when starting asterisk. Needless to say, I can't run asterisk without my 
PRI.


Guess I'll start reverting code backwards day by day until I find the 
code that works.


-Matthew

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Re: [Asterisk-Users] Looking for better "Follow Me"

[Matthew Boehm]

Hauke Zuehl wrote:

Hi everybody :)

I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to my 
private Asterisk. I want to call my wife at home so her SIP phone rings. She 
does not pick up the phone (maybe she is somewhere in the house and has to 
run to the phone) so after 15 seconds her cell phone should ring.

Until now it is a classic "follow me" but what I want:
I want both phones (SIP and cell) ringing and if one phone is picked up the 
other phone should stop ringing.


This is easially accomplished via the following single line:

exten => 3044,1,Dial(SIP/herphone&ZAP/R1d/5551212,60)

Go read about app_dial on the wiki; it has lots of features built into it.

-Matthew

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[Asterisk-Users] Outgoing Context being mistake for dialplan?

[Matthew Boehm]

Here is one that happens at random. Probably 1 out of 1000 calls will show
this:

-- AGI Script Executing Application: (Dial) Options:
(SIP/[EMAIL PROTECTED]|60)
-- Called [EMAIL PROTECTED]
Aug 31 18:27:28 NOTICE[25965]: pbx.c:1681 pbx_extension_helper: Cannot find
extension context 'xocommunications'
-- SIP/xocommunications-a5f5 is ringing

The notice is correct, I have no extension context called
"xocommunications". I do have a SIP Peer called that though.

Somehow it's getting confused.

Anyone else seen this before?

Like I said, I could have 1000 or more calls go out just like this one with
no problems. Just get this one random

-Matthew


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[Asterisk-Users] 1.2beta and PRI and CDR Corruption

[Matthew Boehm]

Anyone out there running 1.2beta with a PRI and having CDR problems?

I just upgraded to most recent everything and now my CDR's look like this:

"","","9035646130","copper_routing","","Zap/65-1","SIP/netl-a3ac","Dial","SIP/[EMAIL PROTECTED]|60","2005-08-31 
13:03:09","2005-08-31 13:03:20", "2005-08-31 
13:03:29",20,9,"ANSWERED","DOCUMENTATION"


That is a direct text copy/paste from Master.csv

This only seems to affect incomming PRI calls. All other calls (inc SIP, 
out SIP, out PRI) show correct CDRs.


I'm worried this corruption of data may eventualty lead to a crash but 
so far nothing.


CallerIDName is parsed correctly:

"","p^","8322008630","macro-faxrecordvoicemail","""BOEHM MATTHEW  "" 
","Zap/2-1","SIP/3044-b976","Dial","SIP/3044|10|wt","2005-08-31 
13:06:36","2005-08-31 13:06:36","2005-08-31 
13:06:42",6,6,"ANSWERED","DOCUMENTATION"


but CallerIDNumber is not; as evidenced above.

Any thoughts? Ideas? Should I be reverting asterisk code backwards or 
libpri?


-Matthew

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Re: [Asterisk-Users] odbc realtime update problem

[Matthew Boehm]

Julian Lyndon-Smith wrote:

After an afternoon of chasing all sorts of dead-ends (permissions etc) I 
finally changed the uniqueid from an int to a character field, and it 
all updates ok now.


Now, is this a problem with res_odbc, the linux odbc client or the sql 
server itself ?


	Must be cause I am using res_config_mysql and my uniqueid field is an 
INT and everything works great.


-Matthew

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Re: [Asterisk-Users] Realtime Queues and Agents

[Matthew Boehm]

Julian Lyndon-Smith wrote:

That's a bugger.

Forgive me for asking, but how is is possible to be able to have SIP 
realtime (adding new sip phones in without having to reload) but we 
can't have agent realtime ?


In my simple mind I substitute agent for SIP and can't compute :)


 because you don't add new sip phones via RealTime. you add them 
outside the realm of asterisk into your database.


-Matthew

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Re: [Asterisk-Users] Realtime Queues and Agents

[Matthew Boehm]

Julian Lyndon-Smith wrote:
We use agents and queues, with CVS HEAD. I've read up on realtime queues 
and queue members, (and actually understand it!) but there is no 
reference to agents.


Is it possible to have realtime agents as well ?

Julian.


	No there isn't. And there won't be until RealTime gets updated with 
'INSERT' and "DELETE" abilities.


-Matthew

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Re: [Asterisk-Users] realtime and include

[Matthew Boehm]

Luca Lafranchi wrote:

I'm interested for this thread, can you explain with an example please?

In my extensions.conf I have

...
[sip.proxy.com]
switch => Realtime/[EMAIL PROTECTED]


in extensions table on mysql I can insert on "app" field the command
"include" and in the "appdata" field my context ?

Luca


Go read:

http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static

-Matthew

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Re: [Asterisk-Users] realtime and include

[Matthew Boehm]

Urban wrote:

Hi,

is there any support for include statement in the database when using 
realtime configurations? I would like to have as much as possible 
configuration in my postgres db but we have different access controls 
for different user contexts (allow international, national etc). Today 
we have different contexts for access rules e.g.

[allow_international]
exten => _00.,1,Dial...

and for users we just include the allow_xxx and deny_xxx contexts. This 
makes it easier since we don't need to change each users dialplan just 
include the right contexts.
Is this possible with realtime? The only way I see is to add/remove 
switch statements in extensions.conf and then we back to make the 
changes in extensions.conf and not in the database...


	If you store the extensions.conf in database, then it will work. If you 
want to use the switch, then no.


-Matthew


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Re: [Asterisk-Users] storing voice messages in DB SQL

[Matthew Boehm]

Its the same syntax for every other config. Just look at every other config
option and replicate.

Odbctable=mytablename

Or

Odbctable => mytablename

-Matthew


> From: harry gaillac <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sun, 28 Aug 2005 12:11:07 +0200 (CEST)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] storing voice messages in DB SQL
> 
> hello,
> 
> According to docs/README.odbcstorage how can we set :
> 
> ///
> The database name (from /etc/asterisk/res_odbc.conf)
> is in the
> "odbcstorage" variable in the general section of
> voicemail.conf.
> You may modify the voicemessages table name by using
> odbctable=??? in voicemail.conf
> ///////
> 
> what's the right syntax in voicemail.conf ?
> Harry
> --- Matthew Boehm <[EMAIL PROTECTED]> a écrit :
> 
>> Yes. Look in the apps/Makefile for USE_ODBC_STORAGE
>> and read in the docs/
>> for a table structure.
>> 
>> Right now it is ODBC only.
>> 
>> -Matthew
>> 
>>> From: harry gaillac <[EMAIL PROTECTED]>
>>> Reply-To: Asterisk Users Mailing List -
>> Non-Commercial Discussion
>>> 
>>> Date: Sat, 27 Aug 2005 16:12:09 +0200 (CEST)
>>> To: 
>>> Subject: [Asterisk-Users] storing voice messages
>> in DB SQL
>>> 
>>> Hello,
>>> 
>>> Can we store voice messages in a database instead
>> of
>>> files.
>>> 
>>> Regards
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>> 
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> 
> 
> 
> 
> 
> 
> 
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Re: [Asterisk-Users] storing voice messages in DB SQL

[Matthew Boehm]

Yes. Look in the apps/Makefile for USE_ODBC_STORAGE and read in the docs/
for a table structure.

Right now it is ODBC only.

-Matthew

> From: harry gaillac <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 27 Aug 2005 16:12:09 +0200 (CEST)
> To: 
> Subject: [Asterisk-Users] storing voice messages in DB SQL
> 
> Hello,
> 
> Can we store voice messages in a database instead of
> files.
> 
> Regards
> 
> 
> 
> 
> 
> 
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Re: [Asterisk-Users] Realtime and database structure

[Matthew Boehm]

 Pablo Ezequiel Fernández wrote:
Is there any official structure for the tables for doing realtime sip and 
extensions ? I see there's a big (big in lot's of fields) on 
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip and I've 
seen other partial tables in other places.

Thanks.


The only fields that are required are:

name
ipaddr
port
regseconds

The rest are all option as per sip.conf. If it is optional in sip.conf, 
then it is option in realtime.


-Matthew

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Re: [Asterisk-Users] realtime sip channel configuration -> insecure option

[Matthew Boehm]

Billy wrote:


 `insecure` varchar(4) default NULL,


	This can be changed. I just read the chan_sip.c code and the following 
values are acceptable:


"very"
"yes"
"true"

"port"
"invite"
"port,invite"
"invite,port"

The varchar(4) was originally intended for: "very", "yes" or "no"

-Matthew

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[Asterisk-Users] OT: Are you using a Lucent?

[Matthew Boehm]

Is anyone out there using Lucent brand equipment to handle an incomming 
DS3, converting all 672 calls to SIP (as G729) and sending those to 
Asterisk/SER over ethernet?


If you are and are willing to speak to my boss about your experiences 
(over the phone) with it, please contact me off list.


We have a possible contract with a local CLEC to handle their long 
distance, and they want to send to us using DS3 and SS7.


I'm trying to convince my boss to use a $9K Lucent, but he wants to 
spend much more by breaking out the DS3 into DS1's and stack up 6 
asterisk boxes with 1 4-port card in each.


Again, if you are using Lucent and are willing to speak to my boss about 
your experiences, please contact me off list so I can setup a call.


Thanks,
Matthew

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[Asterisk-Users] RealWorld Stats; Not achieving expected results

[Matthew Boehm]

Hey guys,
 We have a brand new Dell Poweredge 1850, Single Proc 3Ghz, 2GB RAM, 
15K RPM HD's RAID 1.


 We also have a Sangoma 4 port T1/PRI card.

 We are not using G729. Everything is G711.

 Every call is PRI -> Asterisk G711 -> Sip Carrier

 We just filled up 2 PRI's and reached a CPU usage of 70%.

 There's no way we can do 4 PRI's worth of traffic. Will a 2nd CPU 
really make that much difference? Why is there so much CPU used going 
from PRI to 711? I could understand 70% usage if I was using 729 but I'm 
not.


 I was expecting to be able to process 96 calls (all 4 PRIs) with about 
50% (or less) CPU usage.


 Any clues? Suggestions?

Thanks,
Matthew

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Re: [Asterisk-Users] Sql Realtime

[Matthew Boehm]

Jimmy Smith wrote:


.. options i found would be to use mysql  from the asterisk distro..
but are the memory leaks fixed ?


Options are to use res_config_mysql found in asterisk-addons.

What memory leaks?

-Matthew

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Re: [Asterisk-Users] Lots of console; attach and grep?

[Matthew Boehm]

Eric Wieling aka ManxPower wrote:

You mean like the info in /var/log/asterisk which is configured via 
/etc/asterisk/logger.conf ?


Damn. If I change any logging, that's going to require an asterisk 
restart isn't it?


-Matthew

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[Asterisk-Users] Lots of console; attach and grep?

[Matthew Boehm]

We have recently started routing about 3 PRI's worth of traffic thru our 
asterisk box.


The text on the console now flys by so damn fast, I can't really see 
what the heck is going on. Even with verbosity 0 and debug 0 it is still 
so fast.


Is there some way I can attach to the console in a way that will allow 
me to grep or otherwise filter the text so I can focus on something in 
particular?


Thanks,
-Matthew

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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] me_ext

[Matthew Boehm]

John Novack wrote:

Both modules should be able to use either hostname, IP, or local 
socket. They both use the same mysql API code to connect to the server.



Should perhaps, but they DON'T!


	I will test this but again, they use the same mysql_real_connect() to 
connect to the server.



Why  isn't there a good example of that on the Wiki or in the docs?
If you understand how to do that, then TEACH others!


	It says it quite plainly in all the sample configs that if you are 
running a database on the same machine that you can use sockets.


-Matthew

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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] me_ext

[Matthew Boehm]

John Novack wrote:
I also experienced some difficulty in connecting to mysql, though for 
CDR, using HEAD from 2/24/05 and finally discovered that the 
CDR_mysql.conf wanted the host NAME in hostname, while res_mysql.conf 
wanted the IP address. Both Asterisk and mysql on the same machine, 
running RH9


	Both modules should be able to use either hostname, IP, or local 
socket. They both use the same mysql API code to connect to the server.


	If mysql and asterisk are on the same machine, why aren't you using 
sockets?


-Matthew

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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] me_ext

[Matthew Boehm]

Araba, Michael wrote:
I am using the current HEAD of asterisk and for asterisk-addons. I have 
been trying to setup realtime mysql voicemail but no sucess. I keep 
getting this error below. The necessary modules are loaded,

res_config_mysql.so ...


There is no point to repeating what you just posted.


res_mysql.conf settings
[general]
dbhost = localhost
dbname = RealTimeMaster
dbuser = xxx
dbpass = xxx
dbport = 3306
dbsock = /tmp/mysql.sock


	Does /tmp/mysql.sock exist? I don't think it does since the error you 
get tells you this. Where is your mysql socket? Is mysql even running on 
this machine?



extcconf.conf


	This file looks correct. Make sure the name of the file is correct: 
extconfig.conf


Aug 24 00:56:50 DEBUG[963]: res_config_mysql.c:597 mysql_reconnect: 
MySQL RealTime: Cannot Connect: Can't connect to local MySQL server 
through socket '/tmp/mysql.sock' (2)


This error is quite self-explanatory.

-Matthew

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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED]

[Matthew Boehm]

"cdr_addon_mysql.c:264 my_load_module: Unable to load config for mysql
CDR's: cdr_mysql.conf"


This is a problem. Where are your config files?


I don't have res_mysql.conf in /etc/asterisk/.


	Well, there is your problem again. Why don't you have config files? You 
can't expect res_config_mysql to use the res_config_odbc config file can 
you?


	There are sample configs for both of these modules. How come you didn't 
install them?


-Matthew


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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] me_ext

[Matthew Boehm]

Araba, Michael wrote:


mail*CLI> realtime mysql status
Aug 24 00:56:50 ERROR[963]: res_config_mysql.c:596 mysql_reconnect: 
MySQL RealTime: Failed to connect database server RealTimeMaster on 
localhost. Check debug for more info.
Aug 24 00:56:50 DEBUG[963]: res_config_mysql.c:597 mysql_reconnect: 
MySQL RealTime: Cannot Connect: Can't connect to local MySQL server 
through socket '/tmp/mysql.sock' (2)


Post your res_mysql.conf. Obviously you cannot login to your mysql box 
for some reason. Can you connect to mysql from your shell?


-Matthew

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Re: [Asterisk-Users] Music On Hold + canreinvite=yes

[Matthew Boehm]

Kevin P. Fleming wrote:

Matthew Boehm wrote:

Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk 
can't send audio (the rtp stream) to the phones.



Umm. "DUH!" Yes it can.

When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio 
stream back to itself for precisely that reason.


Hmm..I stand corrected. And now that I think about it, it seems I jumped 
the gun without thinking.


-Matthew

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Re: [Asterisk-Users] asterisk+realtime

[Matthew Boehm]

Kamran Ahmad wrote:

hello

i m using asterisk-1.0.9.


Come on people. Pay attention.

 What does the very first opening paragraph say:

http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime

-Matthew

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Re: [Asterisk-Users] Music On Hold + canreinvite=yes

[Matthew Boehm]

Ronald Voermans wrote:
For canreinvite=yes to work, I think I need to remove the t argument in 
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways 
stay in the middle. I don't want that, so I removed the 't' argument. 
That works. Now, when two UA are calling, Asterisk gets out of the RTP 
stream. However, when removing the 't' argument, the Music On Hold 
doesn't work anymore between these two UA. If I put one UA on hold, 
Asterisk states that it is starting Music On Hold, but the holding party 
doesn't hear the audio stream.


	Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk can't 
send audio (the rtp stream) to the phones.


-Matthew

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Re: [Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension

[Matthew Boehm]

You spelled Voicemailmain wrong somewhere. Or your extensions are not in
sync with the conf file.

Verify that the extensions.conf is correct then 'extensions reload'.

You can also do "show dialplan " to view what is currently loaded
in memory.

-Matthew


> From: Angus Comber <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 20 Aug 2005 19:21:07 +0100
> To: 
> Subject: [Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No
> application 'Voicemailman' for extension
> 
> Does VoicemailMan have to be installed ?  Why not available.  I have setup a
> mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
> up using *97.
> 
> My *97 code in extensions.conf:
> exten => *97,1,Answer
> exten => *97,2,VoicemailMain([EMAIL PROTECTED])
> exten => *97,3,Hangup
> 
> 
> asterisk console:
> Verbosity was 8 and is now 12
> -- Executing Answer("SIP/200-d83a", "") in new stack
> Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
> application 'VoicemailMan' for extension (default, *97, 2)
>   == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a'
> -- Executing Answer("SIP/200-81f6", "") in new stack
> Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
> application 'VoicemailMan' for extension (default, *97, 2)
>   == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-81f6'
> -- Executing Answer("SIP/201-a86c", "") in new stack
> Aug 20 19:00:24 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
> application 'VoicemailMan' for extension (default, *97, 2)
>   == Spawn extension (default, *97, 2) exited non-zero on 'SIP/201-a86c'
> -- Executing Dial("SIP/201-1e08", "SIP/200|20|Ttm") in new stack
> -- Called 200
> -- Started music on hold, class 'default', on SIP/201-1e08
> -- SIP/200-b925 is ringing
> -- Stopped music on hold on SIP/201-1e08
> -- Nobody picked up in 2 ms
> -- Executing VoiceMail("SIP/201-1e08", "su200") in new stack
> -- Playing 'vm-theperson' (language 'en')
> -- Playing 'digits/2' (language 'en')
> -- Playing 'digits/0' (language 'en')
> -- Playing 'digits/0' (language 'en')
> -- Playing 'vm-isunavail' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav49,
> 0x818eb40
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/200/INBOX/msg format: gsm,
> 0x813a7e8
> -- x=2, open writing:
> /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav,
> 0x818ed88
> -- User hung up
>   == Spawn extension (default, 200, 2) exited non-zero on 'SIP/201-1e08'
> -- Executing Answer("SIP/200-4b1a", "") in new stack
> Aug 20 19:01:57 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
> application 'VoicemailMan' for extension (default, *97, 2)
>   == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-4b1a'
> -- Executing Answer("SIP/200-5369", "") in new stack
> Aug 20 19:02:11 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
> application 'VoicemailMan' for extension (default, *97, 2)
>   == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-5369'
> linux*CLI>
> 
> Angus 
> 
> 
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Re: [Asterisk-Users] Realtime sip_buddies "register=>" how?

[Matthew Boehm]

You can store your entire sip.conf using RealTime. That should allow for
register => to work.

-Matthew


> From: Guillermo Krepper <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 20 Aug 2005 13:05:02 +0200
> To: 
> Subject: [Asterisk-Users] Realtime sip_buddies "register=>" how?
> 
> Hi all
>  
>  I've been doing some testing on realtime using mysql, an have a little
> question that could not find the answer to or maybe its not posible at this
> time.
> Is there a way use "register=>.." on a DB using realtime. For the moment I
> use it in sip.conf. It will help me a lot if this could be store on a DB
> somehow.
> 
> commets or sugestions  ?
> 
> thanks
> Billy
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Re: [Asterisk-Users] Out of G.729 Decoder Licenses!

[Matthew Boehm]

Innocent Evil wrote:

Hello,

I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium
website)
SIP user (100) is calling another SIP user (101).
As 101 is not online, my SIP server is redirecting that call to Asterisk.
Asterisk forward it to 101's voice mail box.

SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server
itself.
But when 100 reach at 101's voice mail, I get this:

Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of
G.729 Decoder Licenses!

I didn't get it.
Would anybody please explain it.


Are the licenses installed? Do "show g729" from CLI. You will need a 
g729 license to access asterisk voicemail from a g729 phone.

-Matthew

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Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

[Matthew Boehm]

Asterisk wrote:

I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.  Which
would be best suited for Asterisk and MySQL?

Bart


I use PHP. Love it. Fast, Easy.

-Matthew

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Re: [Asterisk-Users] options for mysql query from dialplan

[Matthew Boehm]

Hi Damon,
 You are basically doing EXACTLY what we are doing right now; except we 
are doing more.


We now have an AGI PHP script that does the following for every call:

- Connect to MySQL over LAN
- If the dialed number begins with 1, strip it.
- SELECT State FROM lcr_lata WHERE NPA = $dial_npa AND NXX = $dial_nxx
- Do some PHP logic to determine if Interstate vs Intrastate
- SELECT rate, address, technology, prefixes FROM lcr_rates
LEFT JOIN lcr_carriers USING(carrierid)
WHERE NPA = $dial_npa AND NXX = $dial_nxx
AND carrier_active = 1 ORDER BY rate ASC;
- Loop thru results.

lcr_rates has 329,530 rows.
lcr_carriers has 8 rows.
lcr_lata has over 150,000 rows.

Everything preforms in real time.

Here is a sample query of a call that just went thru:

SELECT r.Interstate, rc.name, rc.technology, rc.address, rc.prefix FROM 
lcr_rates r LEFT JOIN lcr_carriers rc ON r.CarrierId = rc.id WHERE r.NPA 
= '254' AND r.NXX = '463' AND r.active = 1 ORDER BY r.Intrastate ASC, 
r.NPA DESC, r.NXX DESC


Query took 0.0025 sec.

I don't see how your table with 300K rows is preforming worse than ours. 
You got indexes?


To make this even better, our MySQL server is a Quad P3 500 Mhz machine.

Works great here.

-Matthew

Damon Estep wrote:

I am using realtime mysql for extensions, sip, and voicemail.

Outbound call routing does not really perform well in realtime
extensions due to the high number of rows in the database (300k), so I
can not use it. It appears with my limited knowledge that the query
method is not robust enough for large databases.

Given the fact that I already have realtime and mysql configured, what
are my options for running a mysql query from the dialplan to find the
provider I want to use for outbound. 


I am not looking for a complete solution, just a hint on the best way to
query my existing mysql database from the dialplan.

I have looked at the MySQL command, and there are a lot of notes about
connection closing and other scary stuff? Does it work?

Are there other native options given the fact that realtime is
configured and in use?

The goal is to run a query against a database like this

SELECT provideralias FROM ldproviders WHERE npa = (digits 2 thru 4 of
dialed number) AND nxx = (digits 5 thru 7)

Then take the provider alias returned and
Dial(SIP/[EMAIL PROTECTED],60).

Next step would be to add a loop for multiple providers, starting with
the lowest cost.

Any hints or comments from the pros?

TIA

Damon


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Re: [Asterisk-Users] SNMP for Asterisk

[Matthew Boehm]

Stojan Sljivic - GDS wrote:

Hi Matthew,

Are you using ucd-snmp or net-snmp?

Regards,
Stojan Sljivic 


UCD is deprecated, its should be net-snmp. I don't include any of the 
ucd headers.


-Matthew

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Re: [Asterisk-Users] SNMP for Asterisk

[Matthew Boehm]

harry gaillac wrote:

Matthew

Could you tell us the differences between your project
and ast-ax-snmpd .

Regards 
Harry


Not really many differences at all. It's basically Andrea's code cleaned 
up. His code was scattered across about 10 different files/headers. I 
combined them into one module; cleaned up the MIB to be compliant with 
current spec; tweaked some code here and there.


Right now, I did remove all the zap specific stuff. I will add it back 
in if people want it.


-Matthew

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Re: [Asterisk-Users] SNMP for Asterisk

[Matthew Boehm]

Stojan Sljivic - GDS wrote:

Hi Matthew,

Nice to hear that.
What will be the license type for your SNMP module?
Are you going to include it in the Asterisk CVS or it will be independent
product?

Regards,
Stojan Sljivic 


Same as asterisk. It will most likely be in asterisk-addons.

-Matthew

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Re: [Asterisk-Users] asterisk with odbc

[Matthew Boehm]

Have you looked in the debug log for any errors? Have you checked the 
ODBC connection using Asterisk CLI? I believe if you type 'odbc ?' then 
you will see a command to check the status of a connection.


-Matthew

Kamran Ahmad wrote:

hello

i am trying to use res_odbc for sipuser. my connection
is working. i have checked using isql. even cdr_odbc
is working but i hav problem in res_odbc. i have
created user in sip_buddies table but asterisk is no
getting user from this sip_buddies table. 


/etc/asterisk/extconfig.conf
[settings]
sipusers=>odbc,asterisk,sip_buddies
sippeers=>odbc,asterisk,sip_buddies

/etc/asterisk/res_odbc.conf
[asteirsk]
dsn=>asteriskdsn
username=>voipbilling
password=>voipbilling
pre-connect=>yes


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Re: [Asterisk-Users] Asterisk configuration from database with res_config

[Matthew Boehm]

That wiki page is old, ugly and out of date. There are many like it and 
if I only knew how to delete wiki pages, I would clean it up some.


The easiest way, Frank, to do what you want is to download CVS-HEAD and 
use ARA to store your config files. Also download addons from HEAD and 
you can use the native mysql realtime driver.


http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime

-Matthew

Frank Aartman wrote:

I want to let Asterisk read its configuration from a mysql database. I
configured everything according to the wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config.
However it doesn't work. I am using 1.0.8 asterisk version and here are
my config files:

Extconfig.conf:

[settings]
;uncomment to load queues.conf via the db engine.
;queues.conf =>  odbc,mysql1,ast_config
;extensions.conf =>  odbc,mysql1,ast_config
sip.conf => odbc,mysql1,ast_config

res_odbc.conf:

;;; odbc setup file 


[mysql1]
dsn => MySQL-asterisk
username => blaat
password => blaat
pre-connect => yes
[mysql2]
dsn => MySQL-asterisk
username => myuser
password => mypass
pre-connect => yes

odbc to mysql is working fine, I tested it. here is my odbc.ini from
/etc/

[MySQL-asterisk] 
Description = MySQL Asterisk database

Trace   = Off
TraceFile   = stderr
Driver  = MySQL
SERVER  = localhost
USER= blaat
PASSWORD= blaat
PORT= 3306
DATABASE= asterisk

I used the load_res_config.pl to put the sip.conf into the database in
ast_config. Via phpMyadmin I can see the data in there correctly. When
booting or reloading Asterisk I don't see anything indicating it is
connecting to odbc. I tried removing the sip.conf from /etc/asterisk,
leaving an empty sip.conf, and only leaving the general section of
sip.conf there. Nothing works.

Cheers,

Frank 
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Re: [Asterisk-Users] How "real time" is realtime?

[Matthew Boehm]

Asterisk Supporter wrote:

How "real time" is realtime?  If the extensions.conf is stored in the
database, does * query it row by row or is it "cached"?  In other words,
given the following exerpt:


	If you store the .conf file using realtime, then yes it is cached. If 
you simply use realtime extensions, then no it is not and you can change 
priority 2 while the call is at 1 and have it take effect immediatly.


-Matthew

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Re: [Asterisk-Users] SNMP for Asterisk

[Matthew Boehm]

Stojan Sljivic - GDS wrote:

Hi Harry,

I have tried to install it, but it gives compilation errors on Fedora Core
3.
Also, patch file for Asterisk was not in sync with version of Asterisk I
had, but I managed to apply changes manually.

Did anyone succeeded to run that package on FC3 or Red Hat Enterprise 3?


I am currently working on a new updated SNMP module for Asterisk. It is 
currently in testing stage as I am having a problem with channel listings.


It will support read-only attributes associated with Asterisk such as 
number channels, number of calls, all config file paths, all modules, 
all apps, etc..


I am hoping to have it ready for inclusion in the 1.2 release.

Once my internal testers have proven it worthy, I will release to public 
for more testing.


Watch for it.

-Matthew

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Re: [Asterisk-Users] Voicemail file permissions

[Matthew Boehm]

Tzafrir Cohen wrote:

On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote:


Is there a way around this w/o giving everyone root privileges!


Run asterisk as its own user/group. We do.

-Matthew

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Re: [Asterisk-Users] realtime caching

[Matthew Boehm]

Jimmy Smith wrote:

pruning breaks asterisk on high loads

at least on all 5 of our servers.

all using different versions and custom.


	You should bug report this if you have a backtrace. Kevin and I worked 
on the pruning stuff (well, he coded and i tested) for a while and 
seemedly got it working.


-Matthew

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Re: [Asterisk-Users] realtime caching

[Matthew Boehm]

Damon Estep wrote:


I may have answered my own question, is it true that realtime extensions
are still queried every call, and only chan_sip is effected by
rtcachefriends?

Damon


	True. RealTime Exensions are queried every time. There is no caching of 
extensions.


If you turn on debug log, you can watch each query.

-Matthew

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Re: [Asterisk-Users] realtime caching

[Matthew Boehm]

We have a web interface where users can update their dialplan online
(not in production yet). The web page modifies the mySQL record.

It seems that some options are not re-read when caching is on, for
example, changing the caller ID value in the sip table has no effect
until a reload (or expiration), so at least in some cases rtcahcefriends
makes realtime notsorealtime.


	No. It is doing exactly what it says it will, "cacheing". If you have 
rtcachefriends turned on, when a peer/user registers the info is pulled 
from DB and added to the internal (a la 'in memory') list that chan_sip 
maintains. If you change something in DB after this occurs then your 
changes won't take affect because chan_sip has no need to re-lookup your 
phones info since the info is already present in memory.


	What you can do is use "sip prune realtime " to remove just the 
single peer/user from memory. And you can force a reload of that peer 
from realtime by using "sip show peer  load".


	If you want pure realtime where chan_sip always pulls from db, then 
turn caching off. Keep in mind that turning caching off will remove MWI 
and NAT functionality.


-Matthew

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Re: [Asterisk-Users] realtime caching

[Matthew Boehm]

> I have reviewed the info below from the sip.sample.conf, but I must be
> dense, still don’t get it.



"Do you find the RealTime comments in sip.conf just a little too 
confusing? Are you frustrated by the use of double negatives in 
configuration options? Do not be afraid. You are not alone. Follow the 
path to enlightenment and visit:

 http://bugs.digium.com/view.php?id=4075";

> It is my understating that removing rtcachefriends will break MWI? Is
> that true?

Yes.

  What exactly are you trying to accomplish? Are your peers/users not 
being updated in your database? Are you sure? Are you watching debug for 
SQL log?


-Matthew





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Re: [Asterisk-Users] quad t1 / 1U rack server combos

[Matthew Boehm]

Damon Estep wrote:

What 1u server combos work with the new quad pri cards UNDER LOAD (more 
than 75% channel use). Every user that buys a Digium PRI card should not 
have to play hit or miss with 2 or 3 servers that cost more than the 
card to get it to work.


We use a Sangoma 4 port T1 card in our Dell Poweredge 1850 (1U) and it 
works like a champ.


-Matthew

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Re: [Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco

[Matthew Boehm]

 > I have been doing a bit of this too lately.  This was also useful.


http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html

Dan


	What about for PRI lines? We get echo every now and then. The docs link 
above references FXO lines. We have none. But we do have 4 PRIs.


-Matthew

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Re: [Asterisk-Users] Multiple Asterisk Installations + SER

[Matthew Boehm]

Ronald Voermans wrote:

I'm not sure I understand what you mean...

I want to have internal extensions (100, 101, 102, etc.) and some full
phone-numbers (10 digits). How do I implement this in *? 


Ronald


Right. We have 3 contexts. 1 is for all incomming traffic from PRI or 
other carrier. 1 for each company and 1 for all outbound. For us, each 
company has their own context. This context handles all "extensions" 
local to the company. (eg, 100, 101, etc).


Then you have a pattern match for when the company dials a 10+ digit 
number. (or 9 followed by any number of digits) We send all these 
"outbound" numbers from all company contexts to a master context for 
outboud dialing (using Goto).


This 'master outbound' context actually includes the 'master incomming' 
context as part of its dialplan so if any customer dials the 10 digit 
number of another company, the call stays within asterisk, completly SIP.


If no match is found, the call is sent thru a PHP script for least cost 
routing out via local PRI or SIP carrier.


-Matthew

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Re: [Asterisk-Users] Multiple Asterisk Installations + SER

[Matthew Boehm]

Brian Capouch wrote:

I'm trying to figure out how to do some things like round-robin server 
balancing and the like using Realtime, and it seems like the "right" way 
to do it would be either via pre-processing the SQL requests coming in, 
or using stored procedures in the database that would accomplish the 
same thing.


Not quite sure I understand the need to pre-process SQL requests. We 
just run 1 MySQL server on a RAID5 array. 1 webserver running the 
scripts which allow the customers to login and do their configuring.


-Matthew

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Re: [Asterisk-Users] Multiple Asterisk Installations + SER

[Matthew Boehm]

> Do you also have some kind of tool so the companies can
> manage their own context ?

 Yup. We use RealTime Extensions. Customers login to a password protected
website. User/pass is tied to their context so they are able to
add/delete/update anything in their extensions context.

 We tell them that extensions managing is a highly advanced tool. If they
screw it up, they get charged a fee for us to fix it.

-Matthew


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Re: [Asterisk-Users] Multiple Asterisk Installations + SER

[Matthew Boehm]

> - I've installed multiple asterisk instances on one server (via
> vserver). Each * is for one customer, and has it's own extensions (like
> 100, 101, 102, etc.) Note that the same extension can exist on other *
> instances

This is completely UNNECESSARY if you simply use contexts. We have 1
asterisk server running 6 different companies and a good majority of their
extensions overlap. This is very easy to configure.


-Matthew


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Re: [Asterisk-Users] list in asterisk cli is getting too long

[Matthew Boehm]

Hilton Williams wrote:
- Original Message - From: Ronald Wiplinger To: Asterisk Users 
Mailing List - Non-Commercial Discussion Sent: Friday, August 12, 2005 
6:04 AM

Subject: [Asterisk-Users] list in asterisk cli is getting too long

How can I use something like|morein CLI ?

The lists are getting too long, like   sip show users


	You can also use "sip show peers like " to truncate the list. 
Ex: "sip show peers like 300" will only show peers whos username starts 
with or contains 300.


-Matthew

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Re: [Asterisk-Users] app_voicemail.c still looking for config fileeven I try to configure the voicemail from database.

[Matthew Boehm]

Wei Kun wrote:


mysql> select * from extensions_table;
++--+---+--+---++
| id | context  | exten | priority | app   | appdata|
++--+---+--+---++
|  1 | from-sip | 2000  |1 | Dial  | SIP/2000|20|
|  2 | from-sip | 2000  |2 | Voicemail | u2000  |


You need to call voicemail app with a context. Change the line above to 
show [EMAIL PROTECTED]


-Matthew

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[Asterisk-Users] Cisco 79XX and VLANS

[Matthew Boehm]

Hey gang,
 We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are 
also using all Cisco Switches and Routers. Everything works great except 
that when you reboot a phone it takes like 3-5 minutes for it to come up.


 The phones spend tons of time 'Configuring VLAN..' We don't run any 
VLANs. Is there some way to skip this?


 In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin 
VLAN Id' set to blank values.


 Any ideas?

Thanks,
Matthew

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Re: [Asterisk-Users] Realtime + MYSQL

[Matthew Boehm]

Timur V. Elzhov wrote:


So the correct line in extconfig.conf must be

voicemail => mysql,asterisk,voicemail_users


Yes, Timur is correct. By stating that you want to bind "voicemail.conf" 
you mean you want to store the config file itself. This is not what you 
are looking for. Change the line above to what Timur says and it should 
work fine.


-Matthew

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Re: [Asterisk-Users] app_voicemail.c still looking for config file even I try to configure the voicemail from database.

[Matthew Boehm]

Wei Kun wrote:

Hi
I am trying to make asterisk load config from database, so far I get the
sip, extension working, but voicemail seems still looking for config file,
not from the database.
Aug 11 15:04:08 WARNING[9316] app_voicemail.c: No entry in voicemail config
file for '2001'


How exactly are you calling it? Are you specifying the right voicemail 
context? What did the debug log say?


-Matthew

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Re: [Asterisk-Users] asterisk query mysql problem or bug?

[Matthew Boehm]

Don't use commas as delimiters in database. You must use pipe |. Replace 
your commas and see if that does the trick.


-Matthew

Wei Kun wrote:

Hi;
I have entries as below in DB,

mysql> select * from sip_buddies;
++--+--++-+++---
-++--+--+
| id | name | context  | defaultip  | host| mailbox| type   |
regseconds | ipaddr | username | port |
++--+--++-+++---
-++--+--+
|  1 | 2000 | from-sip | 10.1.2.192 | dynamic | [EMAIL PROTECTED] | friend |
1123733887 | 10.1.2.192 | 2000 | 5060 |
|  2 | 2001 | from-sip | 10.1.2.220 | dynamic | [EMAIL PROTECTED] | friend |
1123733888 | 10.1.1.220 | 2001 | 5080 |
++--+--++-+++---
-++--+--+
2 rows in set (0.01 sec)

mysql> select * from extensions_table;
++--+---+--+---++
| id | context  | exten | priority | app   | appdata|
++--+---+--+---++
|  1 | from-sip | 2000  |1 | Dial  | SIP/2000,20|
|  2 | from-sip | 2000  |2 | Voicemail | u2000  |
|  3 | from-sip | 2000  |  102 | Voicemail | b2000  |
|  4 | from-sip | 2000  |  103 | Hangup||
|  5 | from-sip | 2001  |1 | Dial  | SIP/2001   |
|  6 | from-sip | 2001  |2 | Voicemail | u2001  |
|  7 | from-sip | 2001  |  102 | Voicemail | b2001  |
|  8 | from-sip | 2001  |  103 | Hangup||
|  9 | from-sip | 2999  |1 | VoicemailMain | ${CALLERIDNUM} |
++--+---+--+---++
9 rows in set (0.00 sec)

Somehow the program get the info '2001,20' stripped from extensions_table
appdata column 'SIP/2001, 20', and try to look it up in sip_buddies name
column as debug output below.

Aug 11 12:23:05 DEBUG[23952] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sip_buddies WHERE name = '2001,20'

Of course, it can't find it, and go to second step for voicemail. If I
change the appdata to 'SIP/2001', it can find it and ring remote party, the
problem is it rings for ever without the 20 hint.

Any hints for this problem?

Thanks
Kun

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Re: [Asterisk-Users] realtime odbc/mysql eating connections

[Matthew Boehm]

Since you are using ODBC, this seems more likely to be an ODBC issue. If 
you are concerned, you should just use the native MySQL RealTime driver. 
It does not exibit the behavior you mentioned.


-Matthew

Frank Sautter wrote:
our asterisk is configured to retrieve sippeers and iaxpeers via odbc 
from a mysql database. after each call "show processlist;" within the 
mysql console shows 2 more persistent connections which are showing no 
further activity and will not go away even after restaring asterisk.


is anybody else experiencing this?
what can i do do resolve this?

this is a "show processlist" on the mysql console
+-+--+---+--+-+---+---++ 


|  Id | User | Host  | db   | Command | Time  | State | Info
+-+--+---+--+-+---+---++ 

|   7 | asterisk | localhost | asterisk | Sleep   | 2 |   | NULL 
   |   8 | asterisk | localhost | asterisk | Sleep   | 13596 | 
  | NULL|  11 | asterisk | localhost | asterisk | Sleep 
  | 13596 |   | NULL

. stuff deleted ...
| 171 | asterisk | localhost | asterisk | Sleep   | 31|   | NULL
| 172 | asterisk | localhost | asterisk | Sleep   | 31|   | NULL
| 173 | asterisk | localhost | asterisk | Sleep   | 1 |   | NULL
| 174 | asterisk | localhost | asterisk | Sleep   | 1 |   | NULL
+-+--+---+--+-+---+---++ 


160 rows in set (0.00 sec)

# less /etc/odbc.ini
[asterisk]
Description = MySQL Asterisk database
Trace   = Off
TraceFile   = stderr
Driver  = MySQL
Socket  = /var/run/mysqld/mysqld.sock
Server  = localhost
User= asterisk
Password= 
#Port   = 3306
Database= asterisk

# less /etc/asterisk/res_odbc.conf
[asterisk]
dsn => asterisk
username => asterisk
password => 
pre-connect => yes

# less /etc/asterisk/extconfig.conf
[settings]
iaxusers => odbc,asterisk,iaxfriends
iaxpeers => odbc,asterisk,iaxfriends
sipusers => odbc,asterisk,sipfriends
sippeers => odbc,asterisk,sipfriends
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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

[Matthew Boehm]

Justin Selleck wrote:
Is asterisk 2.0 real?  Running in c#?  I see references to it but cannot 
find it anywhere.  


No. It's not out. 1.2 isn't even out. And thank god its NOT programmed 
in C#.


-Matthew

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Re: [Asterisk-Users] Asterisk and PostgreSQL

[Matthew Boehm]

Bastian Schern wrote:

Hello everybody,

now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to
use PostreSQL instead of MySQL?

Regards
Bastian


If you want to go thru the hassle of installing ODBC and all related 
stuff to run PSQL, sure you can.


-Matthew

P.S. stick with mysql. :)

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Re: [Asterisk-Users] How to config voicemail with mysql?

[Matthew Boehm]

res_mysql.conf controls the RealTime interface driver to MySQL.
cdr_mysql.conf controls the MySQL CDR Addon.

Are you running CVS-HEAD? Have you installed res_config_mysql.so?

What happens when you type "realtime mysql status" ?

Did you look in the debug log for errors?

-Matthew

Wei Kun wrote:

Hi;
I followed the online http://www.onlamp.com/lpt/a/3956 to configure
voicemail. Now it works well with voicemail.conf and store voicemail as
file.

Now I want to try to test out storing voicemail within mysql database, but
nothing is inserted into the table. It seems Asterisk still is following the
viocemail.conf. Do I miss some config?

I follow
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail

In exconfig.conf
voicemail => mysql,asterisk,voicemail_users

And the table is in the database. The CDR can insert fine into this
database, but nothing regarding voicemail.

btw, what the relationship between res_mysql.conf and cdr_mysql.conf, looks
like a lot of duplicated info.

Thanks
Kun



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Re: [Asterisk-Users] mysql sock location

[Matthew Boehm]

Wei Kun wrote:

Hi;
In case your * and mysql are running on the same machine, and you get error
"Failed to connect to mysql database server ..." when using  Asterisk with
Mysql database, check the location of mysql.sock

not /tmp/mysql.sock, but /var/lib/mysql/mysql.sock

Regards
Kun


The location of your mysql.sock is completly configurable when you first 
install mysql.


./configure --with-sock=/some/path/to/mysql.sock

If you run with the defaults, yea it won't goto /tmp/mysql.sock

Most admins don't use default.

-Matthew

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Re: [Asterisk-Users] Asterisk MWI and Realtime

[Matthew Boehm]

turned on there. I do get the Sip Notify when a message is left, just no
stutter tone when picking up the phone. I will also refresh my build as
well, hopefully no big changes.


	The stutter tone will be a phone/ATA specific issue not related to 
Asterisk.


	For instance, I know the Linksys PAP2-NA's give you the option of 
stutter tone, or quick ring, etc..when there are voicemails.


-Matthew

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Re: [Asterisk-Users] Asterisk MWI and Realtime

[Matthew Boehm]

Michael Baird wrote:

I'm testing my asterisk system and the realtime backend. My Asterisk
build is rather aged, 03/18/2005 CVS. I have successfully moved Sip
peers and Voicemail boxes to the realtime database backend and this
works very well except for MWI. I don't seem to be able to get MWI to
work when I store the voicemail information in a database backend, from
a flat file it does work fine. I'm using rtcachefriends=yes for my sip
users, per the WIKI, I'm presuming asterisk can't see these mailboxes,
and therefore can't poll them to send the alerts when necessary. Is
there anything that can be done to make this work properly, short of
going back to a flat file for voicemail.conf?

Regards
Michael Baird


We have been using RealTime and Voicemail for quite some time and have 
no problems getting MWI's.


If your source is that old, it might be why the config option isn't working.

-Matthew

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Re: [Asterisk-Users] Problem with realtime SIP

[Matthew Boehm]

vinod malani wrote:

Hi Guys,

We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using 
MySQL & Asterisk Addons.


1.0.7 is NOT CVS HEAD!

1.0.7 is STABLE and RealTime doesn't work on STABLE!

-Matthew

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Re: [Asterisk-Users] Full T38 sip Faxing now Available

[Matthew Boehm]

Chris Mason (Lists) wrote:





Asterisk will re-invite the ATA and 5300 so the audio is passed 
directly between them.


What happens if the ata is on your private network, behind NAT. Will it 
still reinvite?




If you have a good NAT device (like a PIX) then it should. We got 1 to 
go thru by doing NAT=no and canreinvite=yes on an ATA that was behind a PIX.


We are trying again now to force a static public IP to an ATA and see if 
it works that way.


Right now we can't get the re-invites to happen.

-Matthew

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Re: [Asterisk-Users] Full T38 sip Faxing now Available

[Matthew Boehm]

OK. I called the guy and this is basically the jist:

Get a T38 enabled ATA. There are now many out there with firmware 
upgrades to enable T38.


Get a Cisco 5300 (or some other gateway that supports T38).

Register ATA with asterisk. Turn on reinvites for the UA.

Make a call from the ata and have the call then Dial([EMAIL PROTECTED]).

Asterisk will re-invite the ATA and 5300 so the audio is passed directly 
between them.


*bing* T38 faxing.

What this guy is offering is basically the use of his 5300 at 2 cents a 
minute to terminate. Right now he only has incomming Los Angeles abilities.


-Matthew

Marc Storck wrote:

Do you want to share your knowledge how to get it work???

Regards,

Marc



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Re: [Asterisk-Users] app_dbodbc for asterisk stable 1.09

[Matthew Boehm]

app_dbodbc has been publically deprecated by the author and he isn't updating
it. Functionality provided by ast_data is provided by RealTime. You will need
CVS-HEAD to use RealTime. Or wait a month for 1.2 to come out.

-Matthew

Quoting Umar Sear <[EMAIL PROTECTED]>:


Hi,

Has anyone manage to comile app_dbodbc or ast_data with the latest
stable release (1.09). If so can you give some guidence on howto do it
as I have trouble getting either working.

Umar
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Re: [Asterisk-Users] ast_config not updating voicemail password

[Matthew Boehm]

Bruce Komito wrote:

I've been using realtime to store my voicemail configuration in a mysql
table for several months now, and have had no problems...until today.  A
few weeks ago, I upgraded to the latest CVS and today I noticed voicemail
is not updating the password when the user changes it through option 0.
I'm not sure when this started happening, but I assume it was sometime
after I upgraded.

Has anyone else seen a problem like this, and if so, what's the solution?

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


There was a recent patch to voicemail that removed an incorrect error. 
It delt with changing password with realtime.


Does it say on console that voicemail password was changed? Does Allison 
say it was saved?


-Matthew

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Re: [Asterisk-Users] How can I use MySQL in the dialplan?

[Matthew Boehm]

What the hell? NO!

show application MySql

app_addon_mysql is the name of the module.

load app_addon_mysql.so

-Matthew

Quoting Ronald Wiplinger <[EMAIL PROTECTED]>:


Matthew Boehm wrote:


Ronald_Wiplinger wrote:


I would like to put / get some data from an MySQL database.

I want to use this MySQL database also via a web page.


bye

Ronald



app_addon_mysql or use RealTime.


*CLI> show application app_addon_mysql
Your application(s) is (are) not registered

I want to use it for putting stored speed dial numbers into the per 
phone stored register, ... I guess I cannot get that with realtime 
done!!!



bye

Ronald Wiplinger

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Re: [Asterisk-Users] Full T38 sip Faxing now Available

[Matthew Boehm]

Quoting Michael D Schelin <[EMAIL PROTECTED]>:


Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.

Thanks

Michael D. Schelin
ShellTel
626-814-2354


Biting too. Send info.

-Matthew




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Re: [Asterisk-Users] Music on Hold: CPU Intensive Monster

[Matthew Boehm]

Matt Riddell wrote:

Matthew Boehm wrote:


Any ideas for getting processor usage down on MOH?



Encode it in the format you want to send out would help a little.



How do you encode MP3's into G729?

-Matthew

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Re: [Asterisk-Users] cdr_mysql does not write to mysql db

[Matthew Boehm]

Neal Lawson wrote:
using localhost in you mysql conf should work, make sure you linux  box 
and the loopback interface up and has a a entry in your /etc/ hosts for 
localhost and that your firewall (if you have one setup on  your 
localbox) allows traffic from 172.0.0.1 to 172.0.0.1


Don't you mean 127.0.0.1 ?

Plus, in the MySQL API documentation the use of "localhost" indicates to 
the API that you want to use a socket connection.


If you don't want to use sockets (which you should on local machine), 
the you need to change to an IP to use TCP.


-Matthew

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[Asterisk-Users] Music on Hold: CPU Intensive Monster

[Matthew Boehm]

OK. So I did a test last night. All of asterisk's threads where using 
0.0% CPU.


I made 1 call to our call queue.

CPU jumped to average of 9% and stayed around that for the 2 minutes I 
was in the queue just listening to music on hold.


MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA 
using G729.


Can I reasonably assume that the 9% was decoding the MP3, then encoding 
G729?


I tried using Anthm's RAW format but that actually made things worse.

I tried going back to mpeg321 and asterisk still used the same amount of 
CPU.


Any ideas for getting processor usage down on MOH?

-Matthew

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