Re: [asterisk-users] ooh323 doesn't know what to do when bridging calls

[bilal ghayyad]

Dears;

I am having same problem, that when I place a call from the H323 end point 
(even if it is not added in the ooh323.conf), then asterisk handle the call and 
play the wave file in the default context. Also I added endpoint to the 
ooh323.conf and same thing, it keep goes for default context whatever the 
context placed.

My Asterisk vesion is 1.4.25
My Asterisk add-on version is: 1.4.8

What I have to do to capture the call from the IP Phone and route it using the 
correct context that I configured it in the [ ] of the ooh323.conf? Any 
specific thing need to be done?

Regards
Bilal


--
> > Hi guys,
> > 
> > I'm trying out ooh323 and couldn't bridge ooh323 and
> sip/zap. 
> > I'm using netmeeting and set gateway to my asterisk. 
> > 
> > Here's my CLI dump:
> > 
> >   == Spawn extension (h323, , 1)
> exited non-zero on
> > 'OOH323/(null)-8c76'
> >     -- Executing [9...@h323:1]
> Dial("OOH323/(null)-3074",
> > "Zap/8/604xxx") in new stack
> >     -- Called 8/604xxx
> >     -- Zap/8-1 is ringing
> > [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390
> ast_indicate_data:
> > Unable to handle indication 3 for
> 'OOH323/(null)-3074'
> >     -- Zap/8-1 is ringing
> >     -- Zap/8-1 answered
> OOH323/(null)-3074
> > [2008-07-02 15:49:08] WARNING[21544]:
> chan_ooh323.c:1053
> > ooh323_indicate: Don't know how to indicate condition
> 20 on ooh323c_5
> > 
> > My ooh323.conf:
> > 
> > [general]
> > bindaddr=192.168.1.9
> > h323id=ObjSysAsterisk
> > e164=100
> > callerid=asterisk
> > gatekeeper = DISABLE
> > gateway = yes
> > context = h323
> > disallow = all
> > allow = ulaw
> > dtmfmode = rfc2833
> > 
> > 
> > extensions.conf
> > [h323]
> > Exten => ,1,Dial(Zap/8/604xxx)
> > Exten => ,n,Hangup
> > 
> > 604xxx goes to my cell. it rings fine but no
> audio. After I picked
> > up from cell, netmeeting still shows "watiting for
>  to answer"
> > message.
> > 
> > Any ideas?
> 
> I don't like the look of the (null) in the channel names.
> 
> If what you quoted was the whole of your ooh323.conf file,
> you don't have
> any peer, user or friend sections. Try adding something
> like:
> 
> [h323gw]
> type=friend
> context=h323
> ip=192.168.1.200          (or
> whatever the IP of your remote H323 endpoint is)
> port=1720
> 
> If that still doesn't help, please mention what versions of
> asterisk and
> asterisk-addons you are using.
> 
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: t...@softins.co.uk
> - http://www.softins.co.uk
> Play: t...@mountifield.org
> - http://tony.mountifield.org



  

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Why CDR is recording dst value = h?

[Zeeshan Zakaria]

For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if the caller hangs up first, the
'dst' column is always 'h'. I put a NoOp right in the begining of this macro
to verify it.

Any idea why is this happening and how can I have correct 'dst' value if the
caller hangs up first.

[dialout]
exten => _NXXNXX,s,1,Dial(SIP/XX/${EXTEN},30)
exten => h,1,Macro(hangupcall,${EXTEN},${CDR(accountcode)})

[macro-hangupcall]
NoOp(${CDR(dst)})
Set(dialout_num=${ARG1})
Set(user_id=${ARG2})
ResetCDR(vw);
NoCDR();
Hangup();


-- 
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Philipp Kempgen]

Steve Edwards schrieb:
> On Mon, 13 Jul 2009, Danny Nicholas wrote:
> 
>> Just out of curiousity (haven't got to AEL yet), should these exec's be 
>> re-written as AGI calls?
> 
> An exec is executed when the file is reloaded by Asterisk.

Except that in the case of AEL the #exec is not executed at all. :-)

> An AGI, well, you know :)

Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Philipp Kempgen]

Danny Nicholas schrieb:
> Just out of curiousity (haven't got to AEL yet), should these exec's be
> re-written as AGI calls?

I don't think so. #exec's and AGI() are two entirely different
concepts.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Philipp Kempgen]

Tilghman Lesher schrieb:
> On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote:
>> Philipp Kempgen schrieb:
>> > Is Asterisk supposed to evaluate #exec's in an #include'd file?

> The directive "#exec" is not permitted in an AEL configuration file.

I see, that would explain why it doesn't work. :-)

But in that case it's a documentation issue. The extensions.conf
sample says: "The #exec command works on all asterisk configuration
files." I guess it should read "The #exec command works on all
asterisk *.conf files except for asterisk.conf."

Is there a specific reason not to permit #exec in AEL files?

BTW: That's a good example of something to run in
/etc/asterisk/startup.d/*.sh. Thread:
http://lists.digium.com/pipermail/asterisk-users/2009-May/232318.html
http://lists.digium.com/pipermail/asterisk-users/2009-May/232709.html
The story is that I already have a script which recursively evaluates
#include's and #exec's in AEL files and then writes extensions.ael.
I wanted to get rid of the script because there's no clean way to have
it run automatically before asterisk is about to be started but now I
can't.

Is any *.conf file (which permits #exec) guaranteed to be read before
extensions.ael? It would then be possible to (ab)use an #exec in there
to trigger my generator script (which must not output anything then of
course). extconfig.conf? logger.conf? modules.conf? Ugly workaround
but doable.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Steve Edwards]

On Mon, 13 Jul 2009, Danny Nicholas wrote:

> Just out of curiousity (haven't got to AEL yet), should these exec's be 
> re-written as AGI calls?

An exec is executed when the file is reloaded by Asterisk.

An AGI, well, you know :)

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

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Benny Amorsen]

Zeeshan Zakaria  writes:

> I've just found out that CDR(accountcode) variable can only be 20
> characters long, doesn't matter what size the MySQL column has for it.
>
> I need to increase it to at least 30 characters. Any idea how this can be
> accomplished?

As others have said, that requires recompilation, which is a pain.

However, there is another way, in sip.conf:

[foo]
accountcode=foo
setvar=FANCYLONGACCOUNTCODE=foo

Then in the dialplan:

exten => _X!,1,Set(CDR(fancylongaccountcode)=${FANCYLONGACCOUNTCODE})

Now you just need to set up cdr_adaptive_odbc to map
CDR(fancylongaccountcode) to accountcode in the database, or
alternatively use a database view to accomplish that.

I haven't tested it, but it should work.

Even better would be:

[foo]
accountcode=foo
setvar=CDR(fancylongaccountcode)=foo

but I'm not sure whether that works.

Last concern: Does setvar work even for transfers, like accountcode
does?


/Benny


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] transfer option and pressing #

[Alex Samad]

On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote:
> Alex Samad wrote:
> > Hi
> >
> > I have setup forwarding - xfering - where you press # and then the
> > extension. I add t to the dial cmd.
> >
> > My problem is that when you call something like internet banking they
> > want #, but when # is pressed asterisk gets it instead. is there a way
> > around this ?
> >
> > I haven't been able to get asterisk to listen to flash either 
> >
> >
> > Alex
> >   
> The easiest solution would probably be to look in features.conf and 
> change the option for forwarding to require two consecutive # presses.

actually when into features and change all the options to *
instead of #

> 
> The other option would be to put an explicit dial rule for the numbers 
> that need the # bypass and have them omit T and from the dial command.
> 
> You could also set up a dat abase with a simple web front end for your 
> users to enter numbers that need to have the transfer function bypassed 

this is a home system for now (also testbed)

> and do something like this (I use AEL so this is in AEL Format)
> 
> macro specialDial (ext) {
> if (${DB_EXISTS(bypass/${ext})}) {
>Dial (${TRUNK}/${ext});// Dial without transfer
> } else {
>Dial (${TRUNK}/${ext},,T); // Dial With Transfer
> }
> }
> 
> This is assuming you create a table called "Bypass" in your Asterisk 
> Database and add the number to the database.
> 
> Good luck,
> Brent
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
"See, we love -- we love freedom. That's what they didn't understand. They hate 
things - we love things."

- George W. Bush
08/29/2002
Oklahoma City, OK


signature.asc
Description: Digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] unknown RTP codec 126 ??

[gergis.rasmy]

could anyone  help explaining what does this error mean? 
i get this error when make a video/ audio call from X-lite to Bria prof. phone 

rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26'


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Danny Nicholas]

Just out of curiousity (haven't got to AEL yet), should these exec's be
re-written as AGI calls?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, July 13, 2009 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] #exec in #include'd file

On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote:
> Philipp Kempgen schrieb:
> > Is Asterisk supposed to evaluate #exec's in an #include'd file?
>
> I should probably add that it doesn't work for me in case that wasn't
> obvious.
>
> NOTICE[14143]: ael.flex:878 setup_filestack:   --Read in included file
> /e-globals.ael, 1999 chars ERROR[14143]: ael.y:812 ael_yyerror: 
> File: /e-globals.ael, Line 53, Cols: 8-58: Error: syntax error,
> unexpected word, expecting '='
>
> The line in question is
> #exec "/e-globals.ael.php"
>
> execincludes is enabled.

The directive "#exec" is not permitted in an AEL configuration file.

-- 
Tilghman

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Tilghman Lesher]

On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote:
> Philipp Kempgen schrieb:
> > Is Asterisk supposed to evaluate #exec's in an #include'd file?
>
> I should probably add that it doesn't work for me in case that wasn't
> obvious.
>
> NOTICE[14143]: ael.flex:878 setup_filestack:   --Read in included file
> /e-globals.ael, 1999 chars ERROR[14143]: ael.y:812 ael_yyerror: 
> File: /e-globals.ael, Line 53, Cols: 8-58: Error: syntax error,
> unexpected word, expecting '='
>
> The line in question is
> #exec "/e-globals.ael.php"
>
> execincludes is enabled.

The directive "#exec" is not permitted in an AEL configuration file.

-- 
Tilghman

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] chan_ooh323.so and chan_h323.so

[bilal ghayyad]

Hi All;

If already the chan_ooh323.so existed in the /usr/lib/asterisk/modules/ then I 
can not have chan_h323.so? Because I tried alot, and does not have chan_h323 
(but I already have chan_ooh323).

Regards
Bilal


  

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT - How to indent AEL file

[David Backeberg]

On Thu, Jul 9, 2009 at 3:47 AM, Olivier wrote:
> Hi,
>
> As my extensions.ael is becoming quite long (3000 lines), I'm wondering if
> existing indentation tools
> such as vim, indent, ... could improve its formatting (split long lines into
> several ones, align {}, ..)
> Has anyone tried ?

You may prefer to split it into multiple files, using the 'include' syntax.

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.

[Connor Spiess]

Mark,

Would it be possible for you to send me the messages file in the 
var/log/asterisk directory.
If you want you are welcome to send it to the email address below.

Connor Spiess
Network Specialist
cspi...@idea-ma.com


-Original Message-
From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
Sent: Friday, July 10, 2009 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a 
consultant.

Steve,

Thanks for your thoughts. I am tearing out my last bit of hair on this
one.

We only use sip on our internal network to talk to the 7960s

We are getting drops from no-cell phone hard wired phones too.

Unfortunately There are too many drops for me to let this go. :(

Mark

On Jul 10, 2009, at 12:48 PM, Steve Totaro wrote:

> This is an age old Asterisk (and general telephony) problem.  I
> can't blame it all on Asterisk.
>
> Never thought of the 5ess, filed in my memory bank as this is an age
> old problem.
>
> Too bad it happens with SIP providers and not just the little guys
> but XO for instance.
>
> I hear crackling.  Cell phones drop all the time.
>
> On a bad day I get five dropped cell phone calls a day.
>
> Thanks,
> Steve Totaro
>
> On Fri, Jul 10, 2009 at 12:15 PM, Connor Spiess  ma.com> wrote:
> We had the same problem using a Digium T1 card. We switched the
> coding to from NI2 to 5ess and we haven't dropped a call since.
> You will have to check with your service provider to see if they do
> 5ess.
>
> Connor Spiess
> Network Specialist
>
>
> -Original Message-
> From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
> Sent: Friday, July 10, 2009 10:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas
> and a consultant.
>
> Hello Everyone.
>
> We have:
>
> Asterisk 1.4.21.2
> zaptel-1.4.11
> libpri-1.4.5
> Sangoma A101D Connected to a PRI
> Cicso 7960G phones (About 30 of them)
>
> We have a problem with dropped calls that has going on for a long
> time.  We get up to 5 dropped calls on a bad day. They all seem to be
> incoming calls.
>
> I have a recording of what my users report a dropped call sounds like
> right before it drops
>
> http://www.stepawayfromthecomputer.com/drop.wav
>
> Please have a listen to the recording and tell me what you think it
> means
>
> I am looking for any ideas as to what I should do to track this down.
> I would love a lead on a good consultant who can help fix this.
>
> Mark
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> 
> Mail was checked for spam by the Freeware Edition of No Spam Today!
> The Freeware Edition is free for personal and non-commercial use.
> You can remove this notice by purchasing a full license!
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Mail was checked for spam by the Freeware Edition of No Spam Today!
The Freeware Edition is free for personal and non-commercial use.
You can remove this notice by purchasing a full license!


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up phones

[Ott Rose]

I did " set sip debug on " from the CLI

It doesn't scroll messages like it did on Fri


i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which 
isn't either one of the ips of the asterisk server. then it hung up

i do have a dial tone


i just figured something out after reading my post.


if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to the 
extension and the other phone rings. 

still can't get the 99 to call the asterisk server to work i put in the ips of 
the server but it hangs up right away
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 13 Jul 2009 12:57:59 -0500
Subject: Re: [asterisk-users] setting up phones






















I assume you get a dial tone when you pick
up the handset?If you had a good phone-to-asterisk connection, debug would
show a connection or rejection when you did 99#.

 









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

Sent: Monday, July 13, 2009 12:49
PM

To:
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users]
setting up phones



 

added that line to the extensions.conf file because i
could find a way to add it in the GUI. I put it under the dial plan that i have
selected. i just get a busy signal i tried #99 just 99, *99 nothing works.
debugging isnt showing anything.







From: da...@debsinc.com

To: asterisk-users@lists.digium.com

Date: Mon, 13 Jul 2009 12:12:16 -0500

Subject: Re: [asterisk-users] setting up phones



Most folks (AFAIK) use TFTP to connect to
the Asterisk server.  I personally use HTTP, but that took a few days of
research to figure out.  You’re really only using that protocol for
configuration and log transfers.  The actual lifting is done on a TCP or
UDP connection.  Your posts Friday indicated that Asterisk was up and
“functional” but that you couldn’t make your phones talk to it.  I’m
thinking that instead of trying to dial phone-to-phone, that you should first
make one phone talk to asterisk using this little snippet.

 

-  exten => 99,1,Playback(tt-monkeys)

-  exten => 99,2,Playback(vm-goodbye)

-  exten => 99,3,hangup

 

When you
get your phone where it can dial 99 and get a message, you will be ready to
proceed with P2P talking.

 









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

Sent: Monday, July 13, 2009 12:02
PM

To:
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users]
setting up phones



 

Ok here is what i did. 



reinstalled asterisk (i used the make samples option) and asterisk-gui



in the gui i did the following

 created a dial plans using the defaults. no outgoing dial plans just
local

 crated two users

 logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. 



 Still doesn't work. Should i be able to use the configuration server
settings form the phones web gui. it has the options for tftp, ftp, http,
https. I don't know how this is supposed to be configured. I still don't know
what the problem is and sip set debug off does display any info like it was
lastweek. 





I am just trying to use the gui like you suggestd



> Date: Fri, 10 Jul 2009 14:22:25 -0700

> From: asterisk@sedwards.com

> To: asterisk-users@lists.digium.com

> Subject: Re: [asterisk-users] setting up phones

> 

> On Fri, 10 Jul 2009, Ott Rose wrote:

> 

> > I don't think the GUI is editing the conf files correctly. I am not
sure 

> > I have configure things right. At this point i think i am going to
start 

> > from scratch.

> 

> Yea!

> -- 

> Thanks in advance,

> -

> Steve Edwards sedwa...@sedwards.com
Voice: +1-760-468-3867 PST

> Newline Fax: +1-760-731-3000

> 

> ___

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

> 

> asterisk-users mailing list

> To UNSUBSCRIBE or update options visit:

> http://lists.digium.com/mailman/listinfo/asterisk-users







Windows Live™ Hotmail®: Spread the word when you add celeb
photos to your e-mails. Check
it out.



 







Bing™ brings you health information from trusted sources. Try it now.


_
Bing™ brings you health information from trusted sources. Try it now.
http://www.bing.com/search?q=pet+allergy&form=MHEINA&publ=WLHMTAG&crea=TXT_MHEINA_Health_Health_PetAllergy_1x1___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] #exec in #include'd file

[Philipp Kempgen]

Philipp Kempgen schrieb:

> Is Asterisk supposed to evaluate #exec's in an #include'd file?

I should probably add that it doesn't work for me in case that wasn't
obvious.

NOTICE[14143]: ael.flex:878 setup_filestack:   --Read in included file 
/e-globals.ael, 1999 chars
ERROR[14143]: ael.y:812 ael_yyerror:  File: /e-globals.ael, Line 53, 
Cols: 8-58: Error: syntax error, unexpected word, expecting '='

The line in question is
#exec "/e-globals.ael.php"

execincludes is enabled.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up phones

[Danny Nicholas]

I assume you get a dial tone when you pick up the handset?If you had a
good phone-to-asterisk connection, debug would show a connection or
rejection when you did 99#.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

added that line to the extensions.conf file because i could find a way to
add it in the GUI. I put it under the dial plan that i have selected. i just
get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt
showing anything.

  _  

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 13 Jul 2009 12:12:16 -0500
Subject: Re: [asterisk-users] setting up phones

Most folks (AFAIK) use TFTP to connect to the Asterisk server.  I personally
use HTTP, but that took a few days of research to figure out.  You're really
only using that protocol for configuration and log transfers.  The actual
lifting is done on a TCP or UDP connection.  Your posts Friday indicated
that Asterisk was up and "functional" but that you couldn't make your phones
talk to it.  I'm thinking that instead of trying to dial phone-to-phone,
that you should first make one phone talk to asterisk using this little
snippet.

 

-  exten => 99,1,Playback(tt-monkeys)

-  exten => 99,2,Playback(vm-goodbye)

-  exten => 99,3,hangup

 

When you get your phone where it can dial 99 and get a message, you will be
ready to proceed with P2P talking.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

Ok here is what i did. 

reinstalled asterisk (i used the make samples option) and asterisk-gui

in the gui i did the following
 created a dial plans using the defaults. no outgoing dial plans just local
 crated two users
 logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. 

 Still doesn't work. Should i be able to use the configuration server
settings form the phones web gui. it has the options for tftp, ftp, http,
https. I don't know how this is supposed to be configured. I still don't
know what the problem is and sip set debug off does display any info like it
was lastweek. 


I am just trying to use the gui like you suggestd

> Date: Fri, 10 Jul 2009 14:22:25 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] setting up phones
> 
> On Fri, 10 Jul 2009, Ott Rose wrote:
> 
> > I don't think the GUI is editing the conf files correctly. I am not sure

> > I have configure things right. At this point i think i am going to start

> > from scratch.
> 
> Yea!
> -- 
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

  _  

Windows LiveT HotmailR: Spread the word when you add celeb photos to your
e-mails. Check
  it out.

 

  _  

BingT brings you health information from trusted sources. Try it now.
 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] #exec in #include'd file

[Philipp Kempgen]

Hi,

Is Asterisk supposed to evaluate #exec's in an #include'd file?


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up phones

[Ott Rose]

added that line to the extensions.conf file because i could find a way to add 
it in the GUI. I put it under the dial plan that i have selected. i just get a 
busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing 
anything.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 13 Jul 2009 12:12:16 -0500
Subject: Re: [asterisk-users] setting up phones






















Most folks (AFAIK) use TFTP to connect to
the Asterisk server.  I personally use HTTP, but that took a few days of
research to figure out.  You’re really only using that protocol for
configuration and log transfers.  The actual lifting is done on a TCP or UDP
connection.  Your posts Friday indicated that Asterisk was up and “functional”
but that you couldn’t make your phones talk to it.  I’m thinking that instead
of trying to dial phone-to-phone, that you should first make one phone talk to
asterisk using this little snippet.

 

- 
exten
=> 99,1,Playback(tt-monkeys)

- 
exten
=> 99,2,Playback(vm-goodbye)

- 
exten
=> 99,3,hangup

 

When you
get your phone where it can dial 99 and get a message, you will be ready to
proceed with P2P talking.

 









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

Sent: Monday, July 13, 2009 12:02
PM

To:
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users]
setting up phones



 

Ok here is what i did. 



reinstalled asterisk (i used the make samples option) and asterisk-gui



in the gui i did the following

 created a dial plans using the defaults. no outgoing dial plans just
local

 crated two users

 logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. 



 Still doesn't work. Should i be able to use the configuration server
settings form the phones web gui. it has the options for tftp, ftp, http,
https. I don't know how this is supposed to be configured. I still don't know
what the problem is and sip set debug off does display any info like it was
lastweek. 





I am just trying to use the gui like you suggestd



> Date: Fri, 10 Jul 2009 14:22:25 -0700

> From: asterisk@sedwards.com

> To: asterisk-users@lists.digium.com

> Subject: Re: [asterisk-users] setting up phones

> 

> On Fri, 10 Jul 2009, Ott Rose wrote:

> 

> > I don't think the GUI is editing the conf files correctly. I am not
sure 

> > I have configure things right. At this point i think i am going to
start 

> > from scratch.

> 

> Yea!

> -- 

> Thanks in advance,

> -

> Steve Edwards sedwa...@sedwards.com
Voice: +1-760-468-3867 PST

> Newline Fax: +1-760-731-3000

> 

> ___

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

> 

> asterisk-users mailing list

> To UNSUBSCRIBE or update options visit:

> http://lists.digium.com/mailman/listinfo/asterisk-users







Windows Live™ Hotmail®: Spread the word when you add celeb
photos to your e-mails. Check it out.


_
Bing™ brings you health information from trusted sources. Try it now.
http://www.bing.com/search?q=pet+allergy&form=MHEINA&publ=WLHMTAG&crea=TXT_MHEINA_Health_Health_PetAllergy_1x1___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Zeeshan Zakaria]

I guess I'll shorten the accountcodes for my users. Its a little bit extra
work to modify the PHP code, but it is better than playing around with the
asterisk on the production server.

Thanks for your answers. This information will help me in future.

Regards,

Zeeshan

On Mon, Jul 13, 2009 at 9:31 AM, Steve Edwards wrote:

> On Mon, 13 Jul 2009, Zeeshan Zakaria wrote:
>
> > What if I compile it again but on a test machine and then copy cdr.h
> > over to the production one?
>
> Your question demonstrates a lack of understanding of the process of
> building software.
>
> An "h" file is only used during compilation. What you need are the
> Asterisk executable and any modules that reference the changed manifest
> constant.
>
> Your probability of success is small.
>
> You should build Asterisk on the target host.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up phones

[Danny Nicholas]

Most folks (AFAIK) use TFTP to connect to the Asterisk server.  I personally
use HTTP, but that took a few days of research to figure out.  You're really
only using that protocol for configuration and log transfers.  The actual
lifting is done on a TCP or UDP connection.  Your posts Friday indicated
that Asterisk was up and "functional" but that you couldn't make your phones
talk to it.  I'm thinking that instead of trying to dial phone-to-phone,
that you should first make one phone talk to asterisk using this little
snippet.

 

-  exten => 99,1,Playback(tt-monkeys)

-  exten => 99,2,Playback(vm-goodbye)

-  exten => 99,3,hangup

 

When you get your phone where it can dial 99 and get a message, you will be
ready to proceed with P2P talking.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

Ok here is what i did. 

reinstalled asterisk (i used the make samples option) and asterisk-gui

in the gui i did the following
 created a dial plans using the defaults. no outgoing dial plans just local
 crated two users
 logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. 

 Still doesn't work. Should i be able to use the configuration server
settings form the phones web gui. it has the options for tftp, ftp, http,
https. I don't know how this is supposed to be configured. I still don't
know what the problem is and sip set debug off does display any info like it
was lastweek. 


I am just trying to use the gui like you suggestd

> Date: Fri, 10 Jul 2009 14:22:25 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] setting up phones
> 
> On Fri, 10 Jul 2009, Ott Rose wrote:
> 
> > I don't think the GUI is editing the conf files correctly. I am not sure

> > I have configure things right. At this point i think i am going to start

> > from scratch.
> 
> Yea!
> -- 
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

  _  

Windows LiveT HotmailR: Spread the word when you add celeb photos to your
e-mails. Check it out.
 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up phones

[Ott Rose]

Ok here is what i did. 

reinstalled asterisk (i used the make samples option) and asterisk-gui

in the gui i did the following
 created a dial plans using the defaults. no outgoing dial plans just local
 crated two users
 logged into the web interface with each phone and pointed them to our asterisk 
server. Just the Proxy server and Registrar server. 

 Still doesn't work. Should i be able to use the configuration server settings 
form the phones web gui. it has the options for tftp, ftp, http, https. I don't 
know how this is supposed to be configured. I still don't know what the problem 
is and sip set debug off does display any info like it was lastweek. 


I am just trying to use the gui like you suggestd

> Date: Fri, 10 Jul 2009 14:22:25 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] setting up phones
> 
> On Fri, 10 Jul 2009, Ott Rose wrote:
> 
> > I don't think the GUI is editing the conf files correctly. I am not sure 
> > I have configure things right. At this point i think i am going to start 
> > from scratch.
> 
> Yea!
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

_
Windows Live™ Hotmail®: Spread the word when you add celeb photos to your 
e-mails. Check it out.
http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_WL_QA_HM_celebrity_photos1_072009&cat=celebrity___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] transfer option and pressing #

[Brent Davidson]

Alex Samad wrote:
> Hi
>
> I have setup forwarding - xfering - where you press # and then the
> extension. I add t to the dial cmd.
>
> My problem is that when you call something like internet banking they
> want #, but when # is pressed asterisk gets it instead. is there a way
> around this ?
>
> I haven't been able to get asterisk to listen to flash either 
>
>
> Alex
>   
The easiest solution would probably be to look in features.conf and 
change the option for forwarding to require two consecutive # presses.

The other option would be to put an explicit dial rule for the numbers 
that need the # bypass and have them omit T and from the dial command.

You could also set up a dat abase with a simple web front end for your 
users to enter numbers that need to have the transfer function bypassed 
and do something like this (I use AEL so this is in AEL Format)

macro specialDial (ext) {
if (${DB_EXISTS(bypass/${ext})}) {
   Dial (${TRUNK}/${ext});// Dial without transfer
} else {
   Dial (${TRUNK}/${ext},,T); // Dial With Transfer
}
}

This is assuming you create a table called "Bypass" in your Asterisk 
Database and add the number to the database.

Good luck,
Brent



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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ooh323 and h323, it accept the call even not added in h323.conf

[bilal ghayyad]

Dears;

Now using Asterisk H323 (which coming with Asterisk, I just compiled PWLIB and 
OPENH323), now I am placing a call from the IP Phone, the call comes to 
Asterisk, and it goes to the default context, but did not hear any voice of the 
played wave file.

1) Why Asterisk accepted the call without authentication? At least, it should 
be added to the h323.conf.

2) In case we found the method to let Asterisk authenticate only those 
endpoints that added in the h323.conf file, based on what the authentication 
will be if the IP address is dynamic? Is it based on the [ ] and what that 
consider on the IP Phone? Is it the h323 id?

Regards
Bilal




> Dovid Bender 
> wrote:
> > Stay away form ooh323. It tends to crash Asterisk.
> 
> Much better to use it and fix any bugs that make Asterisk
> crash.
> If everyone stays away from it, they will never get found
> and fixed.
> 
> IME, ooh323 is much leaner and cleaner than the other two
> H323
> implementations, as it doesn't rely on external libraries
> of
> specific versions, and doesn't have a load of C++ stuff to
> call
> them.
> 
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: t...@softins.co.uk
> - http://www.softins.co.uk
> Play: t...@mountifield.org
> - http://tony.mountifield.org



  

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Polarity Reversal Incorrect

[Marcus Vinicius]

Hi,

I have a FXO line in TDM410P card. Over some calls, after a few minutes of 
conversation, a busy tone occurs on the call. The call remains up, the two 
sides of the call is heard normally, but remains a busy tone in the middle of 
the conversation. When this occurs in the log appears the following:

Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Got event Polarity Reversal(17) on 
channel 2 (index 0)
Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Ignore switch to REVERSED Polarity on 
channel 2, state 6
Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Ignoring Polarity switch to IDLE on 
channel 2, state 6
Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Polarity Reversal event occured - 
DEBUG 2: channel 2, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= 
1706049808


Anybody know how to fix this problem?

I tried to add relaxdtmf=yes in zapata.conf but the problem persisted. 

zapata.conf

[channels]
context=default
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=no
cancallforward=no
callreturn=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=-2.0
group=1
callgroup=1
pickupgroup=1
faxdetect=no
immediate=no
musiconhold=default
echocancel=yes
relaxdtmf=yes
context=default
busydetect=no

context=default
signalling=fxs_ks
group=2
channel=>1-3




Thank you.


--
Marcus


  

Veja quais são os assuntos do momento no Yahoo! +Buscados
http://br.maisbuscados.yahoo.com___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] open source call center application for Asterisk

[Steve Edwards]

Un-top-posting...

> On Mon, Jul 13, 2009 at 2:19 PM, ashish chauhan <
> ashishchauhan07...@gmail.com> wrote:
>
>> I am new to asterisk. i like to configure call center using asterisk. 
>> please can anyone tell me open source application to fulfill my 
>> requirement.

On Mon, 13 Jul 2009, Sasa Bobek wrote:

> Truth is you don't need anything more then Asterisk to configure a call
> center

How about enough "clue" to specify the requirements in sufficient detail 
to get a relevant answer? 1 seat or 1,000,000? What features?

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

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] open source call center application for Asterisk

[Sasa Bobek]

Truth is you don't need anything more then Asterisk to configure a call
center

On Mon, Jul 13, 2009 at 2:19 PM, ashish chauhan <
ashishchauhan07...@gmail.com> wrote:

> Dear all,
>  I am new to asterisk.i like to configure call center using
> asterisk.please can anyone tell me open source application to fulfill my
> requirement.
>
> thanks
> Ashish Kumar Chauhan
> M T S ,C D A C Chennai
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Steve Edwards]

On Mon, 13 Jul 2009, Zeeshan Zakaria wrote:

> What if I compile it again but on a test machine and then copy cdr.h 
> over to the production one?

Your question demonstrates a lack of understanding of the process of 
building software.

An "h" file is only used during compilation. What you need are the 
Asterisk executable and any modules that reference the changed manifest 
constant.

Your probability of success is small.

You should build Asterisk on the target host.

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

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Zeeshan Zakaria]

What if I compile it again but on a test machine and then copy cdr.h over to
the production one?

Zeeshan

On Mon, Jul 13, 2009 at 3:58 AM, Alex Balashov wrote:

> Zeeshan Zakaria wrote:
>
> > I've just found out that CDR(accountcode) variable can only be 20
> > characters long, doesn't matter what size the MySQL column has for it.
> >
> > I need to increase it to at least 30 characters. Any idea how this can
> > be accomplished?
>
> You will have to go in the source code and change it.  Go into
> asterisk-x.y.zz.?/include/asterisk/cdr.h:
>
> #define AST_MAX_ACCOUNT_CODE20
>
> Then recompile the entire source tree.
>
> If you didn't install from source, you're out of luck.
>
> If you don't want to modify the source, you're out of luck.
>
> -- Alex
>
> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] open source call center application for Asterisk

[ashish chauhan]

Dear all,
 I am new to asterisk.i like to configure call center using
asterisk.please can anyone tell me open source application to fulfill my
requirement.

thanks
Ashish Kumar Chauhan
M T S ,C D A C Chennai
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Push-To-Talk?

[Jay R. Worthington]

Hi,

is it possible to configure asterisk as a Push-to-task-over-cellular Server?
After reading the spec i was expecting that it is just sip, but i can't get
my Nokia to authenticate, no matter what settings i use.

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Go t SIP response 420 "Bad Extension" back from

[DHAVAL INDRODIYA]

hi all

i have a following setup,

Xlite > Asterisk --> Outbound provider
(Dialout)> Client (mobile,landline phone)

Xlite registered on kamailio,

and Outbound call goes via outbound provider phone can ring properly but
when i picked up phone then it suddently hangup call giving

 SIP response 420 "Bad Extension" back from ${provider IP address}


anyone have idea to solve out this

any help  appreciated.


regards
Dhaval
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ooh323 and h323

[Tony Mountifield]

In article ,
Dovid Bender  wrote:
> Stay away form ooh323. It tends to crash Asterisk.

Much better to use it and fix any bugs that make Asterisk crash.
If everyone stays away from it, they will never get found and fixed.

IME, ooh323 is much leaner and cleaner than the other two H323
implementations, as it doesn't rely on external libraries of
specific versions, and doesn't have a load of C++ stuff to call
them.

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

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Alex Balashov]

Zeeshan Zakaria wrote:

> I've just found out that CDR(accountcode) variable can only be 20 
> characters long, doesn't matter what size the MySQL column has for it.
> 
> I need to increase it to at least 30 characters. Any idea how this can 
> be accomplished?

You will have to go in the source code and change it.  Go into 
asterisk-x.y.zz.?/include/asterisk/cdr.h:

#define AST_MAX_ACCOUNT_CODE20

Then recompile the entire source tree.

If you didn't install from source, you're out of luck.

If you don't want to modify the source, you're out of luck.

-- Alex

-- 
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to Change size of CDR(accountcode) variable?

[Zeeshan Zakaria]

I've just found out that CDR(accountcode) variable can only be 20 characters
long, doesn't matter what size the MySQL column has for it.

I need to increase it to at least 30 characters. Any idea how this can be
accomplished?

-- 
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users