Re: [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found

2010-03-16 Thread John Haigh
Thanks for the help as it got my pointed in the right direction. The
problem was during load where there was no dundi.conf in /etc/asterisk
so this failure was in the asterisk log files.

Thank You,

John Haigh

On Tue, Mar 16, 2010 at 12:03 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
 Klaus Darilion wrote:
 These commands are also available for 1.4. Looks like the DUNDI module
 is not loaded. Watch debug logging during module load for errors.

 Try ldd /usr/lib/astersik/modules/res_dundi.so and watch for
 unresolved dependencies.

 Just to be clear here, I think you mean pbx_dundi.so

 Leif.

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[asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found

2010-03-15 Thread John Haigh
Are there DUNDi CLI commands for Asterisk 1.4? I have searched google
and I only see the dundi commands in Asterisk 1.6, although I see
reference to them in older version's of Asterisk such as Asterisk 1.4
here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I
view the CLI commands through help I don't see any of the dundi
commands and there are errors when I run a command such as dundi debug

asterisknowdev5*CLI dundi debug
No such command 'dundi debug' (type 'help dundi debug' for other
possible commands)

Here is my version output.

asterisknowdev5*CLI show version
Asterisk 1.4.24 built by root @ localhost.localdomain on a i686
running Linux on 2009-03-20 21:27:25 UTC

I can reload dundi by doing the following: module reload pbx_dundi.so,
but is there a way I can get these commands?

Thanks,

John Haigh

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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

2004-01-04 Thread John Haigh
This is my favourite response to this post RUN... don't walk..

WELL SAID!

John Haigh

- Original Message - 
From: asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 4:24 PM
Subject: Re: [Asterisk-Users] New to asterisk? RUN... don't walk.





 Here's the deal:
 Asterisk is free. If we go with * we will save $50k.
 It does almost anything. I can make it open my garage door. My
 installation records all conversations and then archives them as
timestamped
 stereo MP3s. Our VB windows application can dial out with a click. All for
 free.
 It's not done. We are not at v1.0. Mr Spencer is a busy guy.
 It might not solve 'your' problem. We contracted the
AgentCallbackLogin
 Queue stuff. That part works great. If you want it modified or fixed, pay
 for it or do it yourself.
 If you change your own oil, do your own plumbing, have more that 3
 computers at home, or have [EMAIL PROTECTED] running, you are either a
 do-it-yourselfer or a geek. Asterisk might be for you. On the other hand,
if
 you can't change a lightbulb or don't know what a dipstick is and have
lots
 of money, then pay someone for a phone system.

 But please stop whining. I have 3 kids. Gettin' tired of it.

 Good day.


 - Original Message - 
 From: Me [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, December 31, 2003 2:37 PM
 Subject: [Asterisk-Users] New to asterisk? RUN... don't walk.


  As a newcomer to Asterisk, you will not be welcomed
  with open arms.  First, you will find almost no
  documentation on it's features.  Second, if you try to
  ask questions, you will be flamed and pointed to
  worthless how-tos and 'the wiki'.  These worthless
  documents can only be useful for explaining how things
  work to those already in-the-know.  Lastly, Asterisk
  is so bug ridden, expect frequent segmentation faults.
   With a community so 'anti-n00b', don't expect your
  problems to be fixed anytime soon.
 
  RUN!!! Don't walk... away from Aterisk.
 
  __
  Do you Yahoo!?
  Find out what made the Top Yahoo! Searches of 2003
  http://search.yahoo.com/top2003
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[Asterisk-Users] Listen to a Call

2003-10-29 Thread John Haigh



Hi,

Is it possible to listen in on an existing call 
already, say between the caller and callee? So the 3rd person would listen to 
the caller and callee on an existing channel without theknowledge of 
either the existing caller and callee.
Thanks,

John Haigh


Re: [Asterisk-Users] Anyone got VM2 working with MySQL?

2003-10-27 Thread John Haigh
I am having the same problem. Here are my findings:

In asterisk/messages log file:

Oct 25 10:55:11 WARNING[19474]: File app_voicemail2.c, Line 2388
(vm_execmain): Couldn't read username

CLI debug output is as follows when accessing the VoiceMailMain2 from
extension 8500:

Executing VoiceMailMain2(SIP/2205-3df0, ) in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0' (context = any)
-- Playing 'vm-incorrect'
-- Playing 'vm-password'
-- Incorrect password '2421' for user '2205' (context = any)
-- Playing 'vm-incorrect'

I added the context in to my exten Voicemail2 statement in extensions to see
if this would help: Voicemail2([EMAIL PROTECTED]) but it did not.

John

- Original Message - 
From: Philipp von Klitzing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 8:11 AM
Subject: Re: [Asterisk-Users] Anyone got VM2 working with MySQL?


 Hi!

  MySQL CDR logging is installed and working..

 Same question, same situation for me.

 Can you connect via localhost  socket for CDR? That didn't work for me
 (on two machines), I need to use hostname  port.

 In voicemail.conf, however, there is no paramter to specify a port (or
 socket), at least not from what I read here on the list.

 How is this supposed to work anyway - with another #include file, or does
 Asterisk internally merge mySQL voice mailbox settings and
 voicemail.conf? When I dial my test setup defined in mySQL then *
 complains that there is no such entry... :-(

 By the way: With Voicemail2 and the web interface is forwarding working
 for you? Appears to be broken...

 Cheers, Philipp


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[Asterisk-Users] Voicemail.conf in MySQL is not functioning

2003-10-26 Thread John Haigh
Hi,

Here is the error from Asterisk messages log file that I forgot to put in.

Voicemail.conf in MySQL is not functioning where I get the following error
from Asterisk messages log file:

Oct 25 10:55:11 WARNING[19474]: File app_voicemail2.c, Line 2388
(vm_execmain): Couldn't read username

CLI debug output is as follows:

Executing VoiceMailMain2(SIP/2205-3df0, ) in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0' (context = any)
-- Playing 'vm-incorrect'
-- Playing 'vm-password'
-- Incorrect password '2421' for user '2205' (context = any)
-- Playing 'vm-incorrect'

Here are my configs

In extensions.conf I am using Voicemail2 and VoiceMailMain2 that has support
for MySQL

exten = 8500,1,VoiceMailMain2

In voicemail.conf I have the MySQL connectivity settings in [general]

dbhost=localhost
dbname=asterisk
dbuser=someuser
dbpass=somepass

As well in voicemail.conf I have commented out the entire [default] section,
and mailboxes.

I do have MySQL working with CDR MySQL from asterisk-addons

thanks,

John Haigh

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RE: [Asterisk-Users] cdr_mysql.so

2003-10-25 Thread John Haigh

- Create a conf file called cdr_mysql.conf in /asterisk/

[global]
hostname=localhost
dbname=asterisk
password=somepass
user=someuser

- Add load = cdr_addon_mysql.so to /asterisk/modules.conf

If you are getting errors with mysql cdr while loading asterisk, check that
you can connect to MySQL from a shell (mysql -u username -p) or from MySQL
Front by entering the login info for your asterisk db user. You may want a
MySQL client so you can view your cdr details until you get a frontend going
(astweb ??).

John Haigh


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, October 25, 2003 1:58 PM
To: Eric Wieling
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cdr_mysql.so


Thanks a lot your earlier suggestion worked. The system was lacking
zlib-devel. Now where do I insert the lines for it to load the
cdr_mysql.so since I have it built?  Can you give me an exact example of
what to put here?
AJ


On Sat, 25 Oct 2003, Eric Wieling wrote:

 gzip is not zlib.  On my Mandrake 9.2 system the zlib packages are:

 zlib1-1.1.4-8mdk
 zlib1-devel-1.1.4-8mdk


 On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote:
  Yes I do have gzip installed on that box.  Any other ideas?
 
  On Sat, 25 Oct 2003, WipeOut wrote:
 
   [EMAIL PROTECTED] wrote:
  
   Yes I do have the mysql and mysql-devel packages installed.  With my
very
   limited knowledge of C/C++ here's what seems to be the culpret line
right
   before the error:
/usr/bin/ld:  cannot find -lz
   Any suggestions here?
   AJ
   
   
   
   
   I think thats gzip.. Have you got gzip installed?
  
 
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[Asterisk-Users] Voicemail.conf in MySQL is not functioning

2003-10-25 Thread John Haigh




Voicemail.conf in MySQL is not functioning where I get the following error 
from Asterisk messages log file:
CLI debug output is as follows:
Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0' (context = any)
-- Playing 'vm-incorrect'
-- Playing 'vm-password'
-- Incorrect password '2421' for user '2205' (context = any)
-- Playing 'vm-incorrect'
Here are my configs
In extensions.conf I am using Voicemail2 and VoiceMailMain2 that has support 
for MySQL
exten = 8500,1,VoiceMailMain2
In voicemail.conf I have the MySQL connectivity settings in [general]
dbhost=localhost
dbname=asterisk
dbuser=someuser
dbpass=somepass
I have commented out the entire [default] section and it's mailboxes. 
I do have MySQL working with CDR MySQL from asterisk-addons
thanks,
John Haigh



[Asterisk-Users] mysql-vm-routines.h setup for MySQL

2003-10-24 Thread John Haigh



Hi,

How do you setup * mysql voicemail conf in MySQL. 
There is a header file, but no config or README as to how to setup MySQL for 
storage of the voicemail.conf in a MySQL database table.

Could someone point out how or the author email so 
that I may contact them directory to set this up. I will send my findings to 
this list after.

Thanks,

John Haigh


RE: [Asterisk-Users] Starting * with G729 licences

2003-10-16 Thread John Haigh




look in the asterisk for a directory called redhat. There is a 
startup script in there.

John


  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of CW_ASN - 
  GusSent: Thursday, October 16, 2003 5:10 PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Starting * 
  with G729 licences
  Hi all:
  
  I've just purchase some licences of G.729 codecs, 
  and I like to bring up * using /etc/rc.d/init.d script. 
  
  Does anyone knows how to start in the "old" 
  way?
  
  
  Thanks in advance,
  
  Gus
  
  
  
  
  



[Asterisk-Users] Integrate Asterisk with Meridian phone system

2003-07-23 Thread John Haigh

Has anyone integrated Asterisk for Voicemail with a Nortel Meridian
phone system or know if this could be done?

Thanks,

John Haigh

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[Asterisk-Users] Asterisk Call Manager doc

2003-07-09 Thread John Haigh

I am looking for a doc out there that on how to use the Asterisk Call
Manager. Can someone let me know what the URL to this is.

Thanks,

John Haigh

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RE: [Asterisk-Users] Asterisk Sacrifice?

2003-07-04 Thread John Haigh

Hi Here is my sip configuration with fwd. I would recommend getting a
fwd account (fwd.pulver.com) as it is free.

;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
; Address to bind to
context = default   ; Default for incoming calls
tos=reliability

register=37526:[EMAIL PROTECTED]/37526

[fwd]
type=friend
secret=mypassword
username=37526
host=fwd.pulver.com

My extensions.conf

; at the top you will find the globals section...specify a variable
called PHONE1 for sip/fwd..then 
; you can change it out or add another phone variable if you have other
sip phones. 

[globals]

PHONE1=SIP/fwd 

; Extension 1234
; this will dial my soft sip phone x-lite when someone dials 1234...no
ever does though
exten = 1234,1,Playback(transfer,skip) ; Please hold while...
exten = 1234,2,Macro(stdexten,1234,${PHONE1})
exten = 1235,1,Voicemail(u1234); Right to voicemail
exten = 1236,1,Dial(${PHONE1},30)   ; Ring forever
exten = 1236,2,Voicemail(u1234); Unless busy


Also check out John Todd's asterisk conf files. I think they are great
and they helped me get my head around all this. 
http://www.loligo.com/asterisk/current/extensions.conf

John Haigh

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kelly
McDonald
Sent: Friday, July 04, 2003 8:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Sacrifice?


bk, 

I'm a newbie myself, but I have at least got * working with a sip
provider, although the quality was not to my liking, I was hooking up
with iconnecthere. Here's what I had

in sip.conf:

[iconnecthere]
type=friend
insecure=yes
port=5060
username=xyz
secret=abc
host=natrelay.deltathree.com
dtmfmode=inband
callerid=15408675512
nat=yes

in extensions.conf:

exten = 8500,1,Dial(SIP/[EMAIL PROTECTED])


This was just a test so I could dial 8500 and it would call my home
phone.

Probably have stuff wrong, but it seemed to work.

For the rest, extensions.conf has enough stuff in it that you can go and
make up your own stuff.

HTH,
Kelly

On Fri, 2003-07-04 at 08:23, BK [address only for mailing lists] wrote:
 Hi
 
 is there any ritual sacrifice a newbie has to perform to be welcome on
 this list?
 
 I am new to this whole PBX thing in general and Asterisk in 
 particular.
 I had hoped that the community on this list would welcome a newbie
like 
 myself and help me with some answers to my stupid questions, but
somehow 
 it seems to me that nobody likes to respond to somebody who appears to

 be a complete beginner -- too much bother and a risk to have to
explain 
 everything from scratch -- better not answer at all and all that.
 
 Well, it may appear that way, but I am not a complete idiot. I know a
 lot about mobile switching centres, HLRs, VLRs, IN service nodes, 
 mediation devices, billing and settlement systems etc -- I just don't 
 know much about PSTN and PBXes. I would appreciate it if somebody
could 
 help me out with a few hints on how to set up my Asterisk box, in 
 particular in respect of VoIP as per my last posting.
 
 thank you very much in advance
 kind regards
 bk
 
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RE: [Asterisk-Users] Is there any real asterisk documentation ?

2003-07-03 Thread John Haigh
Here are a list of howto's. There are not found on the Asterisk site.
You will find most of your help in the mailing list archive. 

http://asterisk.gnuinter.net/ - this howto has links to other Howto's
http://asterisk.drunkcoder.com/ http://asterisk.drunkcoder.com/apps.html
http://asterisk.drunkcoder.com/modules.conf

AGI documentation:
http://home.cogeco.ca/~camstuff/agi.html

I have heard reading the code can help, although I haven't done so yet,
but my sense there is a lot in the code they will help.

John Haigh


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, July 03, 2003 10:36 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is there any real asterisk documentation ?


Hi users

my first week with asterisk and digium boards has been
a little bit hard ($£§!?)

So, please :
where did you find the documentation which gave you
the magics ?
ex :
- what are the events the logger can handle ?
- where is the internal application list ?
- what is each application the role in asterisk ?
- does the dynamic loader scan /modules directory or
is there a modules list somewhere ?
- what is the syntax for modules.conf ?
- where are the parameters for each external module ?
- what is the minimum list for asterisk to run ?
- is there any interdependance between modules and int. applications
?
- etc... etc...

Are the pseudo-howto's listed on the site the existing knowledge
repository ? Do I have to read the comments in source code ? Or just
experiment ?

Thank you per advance for any suggestion !!!
Frank

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