[asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Olivier
Hi,

I don't understand why a SIP hardphone can't provision itself using tftp.
I'm very suspicious about my tftp daemon but I lack basic knowledge of Linux
CLI to pinpoint what's going wrong and separate what belongs to SIP phone
configuration from what comes from tftp server.

What I would like to do is to add a given file in current /srv/tftp
directory and test by hand that tftpd would serve this file when asked.

Googling and reading man pages, I could find command that starts tftp server
but I couldn't find any command asking for a given file (please, note I
don't imply such commands don't exist but maybe, some more general network
tools (netcat, ...) are the one to use for basic tftp testing).

So let's say, you've got :
a perfectly running tftp server somewhere on your LAN,
it holds foo.txt file in its /srv/tftp directory.


Which command could you type in for a LAN workstation to receive this
foo.txt ?

Regards

PS1: I'm using Debian etch
PS2: With nestat -a, I've got this :

Proto Recv-Q Send-Q Local address  Remote addressStatus
udp0  0 *:tftp  *:*

So, I think my tftp daemon is not listening but that's another topic.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Sigma Networks
Olivier wrote:
 Hi,

 I don't understand why a SIP hardphone can't provision itself using tftp.
 I'm very suspicious about my tftp daemon but I lack basic knowledge of 
 Linux CLI to pinpoint what's going wrong and separate what belongs to 
 SIP phone configuration from what comes from tftp server.

 What I would like to do is to add a given file in current /srv/tftp 
 directory and test by hand that tftpd would serve this file when asked.

 Googling and reading man pages, I could find command that starts tftp 
 server but I couldn't find any command asking for a given file 
 (please, note I don't imply such commands don't exist but maybe, some 
 more general network tools (netcat, ...) are the one to use for basic 
 tftp testing).

 So let's say, you've got :
 a perfectly running tftp server somewhere on your LAN,
 it holds foo.txt file in its /srv/tftp directory.


 Which command could you type in for a LAN workstation to receive this 
 foo.txt ?

 Regards

 PS1: I'm using Debian etch
 PS2: With nestat -a, I've got this :

 Proto Recv-Q Send-Q Local address  Remote addressStatus
 udp0  0 *:tftp  *:*

 So, I think my tftp daemon is not listening but that's another topic.
 

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

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
you'll want to turn on logging for tftpd so you can monitor the messages:
tail -f /var/log/messages

to test tftp if the daemon is running on the local server:
tftp localhost
get foo.txt


good luck.


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Ex Vito

 So let's say, you've got :
 a perfectly running tftp server somewhere on your LAN,
 it holds foo.txt file in its /srv/tftp directory.

 Which command could you type in for a LAN workstation to receive this
 foo.txt ?


  tftp is the client, do you have it installed ?... example:

  # tftp hostname
  tftp get /srv/tftp/foo.txt
  tftp ^D
  # cat foo.txt
  ...

  Things to check: is /srv/tftp the tftp directory or is it the os filesystem
  directory where the tftp root resides ?

  Also, the tftp daemon in CentOS is started by xinetd and can be
  invoked with extra -v flags so as to increase logging verbosity.
  Check your dist. This may help...

  Cheers,
--
  exvito

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Olivier
2008/7/27 Ex Vito [EMAIL PROTECTED]

 
  So let's say, you've got :
  a perfectly running tftp server somewhere on your LAN,
  it holds foo.txt file in its /srv/tftp directory.
 
  Which command could you type in for a LAN workstation to receive this
  foo.txt ?
 

   tftp is the client, do you have it installed ?... example:

  # tftp hostname
  tftp get /srv/tftp/foo.txt
  tftp ^D
  # cat foo.txt
  ...


That was exactly what I was after : I installed tftp on my Ubuntu system and
checked Debian tftp server



  Things to check: is /srv/tftp the tftp directory or is it the os
 filesystem
  directory where the tftp root resides ?

  Also, the tftp daemon in CentOS is started by xinetd and can be
  invoked with extra -v flags so as to increase logging verbosity.
  Check your dist. This may help...

Yes, that's the next step.
I could see a tftp service is running ok on my server and I need to increase
its logs to pinpoint root cause.

Thanks for all



  Cheers,
 --
  exvito

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

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 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 --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Olivier
2008/7/27 Sigma Networks [EMAIL PROTECTED]

 Olivier wrote:
  Hi,
 
  I don't understand why a SIP hardphone can't provision itself using tftp.
  I'm very suspicious about my tftp daemon but I lack basic knowledge of
  Linux CLI to pinpoint what's going wrong and separate what belongs to
  SIP phone configuration from what comes from tftp server.
 
  What I would like to do is to add a given file in current /srv/tftp
  directory and test by hand that tftpd would serve this file when asked.
 
  Googling and reading man pages, I could find command that starts tftp
  server but I couldn't find any command asking for a given file
  (please, note I don't imply such commands don't exist but maybe, some
  more general network tools (netcat, ...) are the one to use for basic
  tftp testing).
 
  So let's say, you've got :
  a perfectly running tftp server somewhere on your LAN,
  it holds foo.txt file in its /srv/tftp directory.
 
 
  Which command could you type in for a LAN workstation to receive this
  foo.txt ?
 
  Regards
 
  PS1: I'm using Debian etch
  PS2: With nestat -a, I've got this :
 
  Proto Recv-Q Send-Q Local address  Remote addressStatus
  udp0  0 *:tftp  *:*
 
  So, I think my tftp daemon is not listening but that's another topic.
  
 
  ___
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
  AstriCon 2008 - September 22 - 25 Phoenix, Arizona
  Register Now: http://www.astricon.net
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 you'll want to turn on logging for tftpd so you can monitor the messages:
 tail -f /var/log/messages


tftp is started with this (in inetd.conf) :
tftpdgram   udp waitnobody  /usr/sbin/tcpd
/usr/sbin/in.tftpd /srv/tftp

Thanks to manual tftp  (tftp foo.txt) I could find that this tftp server
defaults its log to syslog (grep foo.txt /var/log)

Now, I'm certain :
- a tftp server is somehow running
- I can't see any tftp request/reply from IP phone in /var/log/syslog

Maybe, DHCP server is forwarding IP phone to a wrong TFTP server address.
Anyway, that's another topic.

Thanks for helping.



 to test tftp if the daemon is running on the local server:
 tftp localhost
 get foo.txt


 good luck.


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

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 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 --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Visual Dial Plan

2008-07-27 Thread Peter Lindquist

Dean Collins wrote:


I just stumbled across this on youtube.

 


Does any on the list us it? This is the first I've heard over it.

 


http://www.youtube.com/watch?v=H1j5OrgL1og

 

 


Regards,

Dean Collins
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

+1-212-203-4357 (New York)
+61-2-9016-5642 (Sydney)
http://www.Cognation.net http://www.Cognation.net/profile

 

Yes, I use it and it is a great tool I think. If anything I do miss an 
ability to print out the graphical representation of the dial plans in 
the current version - this is being worked on though.


Best regards,

Peter Lindqvist
Voxion Ltd.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] OT - How to test tftp for phones provisioning

2008-07-27 Thread Tzafrir Cohen
On Sun, Jul 27, 2008 at 11:00:51AM +0200, Olivier wrote:
 Hi,
 
 I don't understand why a SIP hardphone can't provision itself using tftp.
 I'm very suspicious about my tftp daemon but I lack basic knowledge of Linux
 CLI to pinpoint what's going wrong and separate what belongs to SIP phone
 configuration from what comes from tftp server.
 
 What I would like to do is to add a given file in current /srv/tftp
 directory and test by hand that tftpd would serve this file when asked.
 
 Googling and reading man pages, I could find command that starts tftp server
 but I couldn't find any command asking for a given file (please, note I
 don't imply such commands don't exist but maybe, some more general network
 tools (netcat, ...) are the one to use for basic tftp testing).

First and foremost, 

  netstat -lup | grep tftp

or (probably faster):

  netstat -lnup | grep 69

This will show you the name of the process that listens on the standard
TFTP (UDP) port, if any.

Debian includes 3 (or maybe 4) tftp servers:

tftpd - Old and probably less maintained version
tftpd-hpa - A newer version. Slightly larger but this is probably the
one to use.
atftpd - Yet another tftpd. Likewise, better than tftpd .

Both atftpd and tftpd-hpa can run either as stand-alone daemons or from
inetd. Check /etc/default/package-name and the docs directory for the
details. IIRC the scripts of the package already run update-inetd for
you.

Another option is to use dnsmasq. If you already happen to use it as a
DNS and/or DHCP server, then you might have noticed the extra TFTP
serving it has added in its latest versions. It only supports read-only
access. But you don't really want to support that insecure tftp upload,
right?

Logs from all the above go to /var/log/daemon.log .

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


[asterisk-users] HASH, HASHKEYS, ClearHash explanation

2008-07-27 Thread Pavel Jezek
Hi, can somebody explain how to use this func/apps in asterisk?
I tried to find some examples on mailinglists or wiki, however without 
success. thanks
PJ

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] Visual Dial Plan

2008-07-27 Thread Matt Watson
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com

--
Matt
http://www.mattgwatson.ca


On 7/27/08, Peter Lindquist [EMAIL PROTECTED] wrote:

  Dean Collins wrote:

  I just stumbled across this on youtube.



 Does any on the list us it? This is the first I've heard over it.



 http://www.youtube.com/watch?v=H1j5OrgL1og





 Regards,

 Dean Collins
 [EMAIL PROTECTED]

 +1-212-203-4357 (New York)
 +61-2-9016-5642 (Sydney)
 http://www.Cognation.net http://www.cognation.net/profile



 Yes, I use it and it is a great tool I think. If anything I do miss an
 ability to print out the graphical representation of the dial plans in the
 current version - this is being worked on though.

 Best regards,

 Peter Lindqvist
 Voxion Ltd.


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

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 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 --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] IAX to work on two ports: 4569 and 4570

2008-07-27 Thread MFH
Depending on how smart your router configuration is, you can leave both 
boxes A and B on port 4569 and then just set up two rules on your router:

Port 4569 - Box A port 4569
Port 4570 - Box B port 4569

If your router is not complex enough to allow a port mapping such as the 
second line above then you'll need to get box B IAX on port 4570 and I 
think people have described how to do that.


bilal ghayyad wrote:
 Again, I did not understand.

 How the router will know if the call that came from Asterisk Box C need to be 
 routed for Asterisk Box A or Asterisk Box B?

 Note: Asterisk A and B both behind DSL router, while Asterisk C in remote 
 side, both sides are connected via Internet.


 Regards
 Bilal

 --

 On Friday 25 July 2008 10:58:28 bilal ghayyad wrote:
   
 The reason that I need to do this is:

 I will have two Asterisk PBX's, and I need both of them to use same
 Internet (so both of them will be behind NAT under same DSL router), in
 that case, how I will distinguish on the router the calls that need to be
 send for box A and the calls that need to send for box B?

 Asterisk C will be in remote side, and will communicate with Asterisk A and
 B, so I was look to have two ports on for IAX to be running on Asterisk C,
 for example: 4569 and 4570, so when box C need to talk with box A, then it
 sends via 4569 and when it needs to talk with box B then it sends via 4570
 port, and DSL router does the routing based on these ports.

 Is there any other solution?
 

 Oh, for outbound traffic, you don't need to define other ports.  Simply
 forward port 4570 on your router to port 4569 on Asterisk B.  Remember that
 unlike SIP, IAX2 does not encode the port number into the protocol, so it'll
 work perfectly fine this way.

 If you're bound and determined to listen to port 4570 on Asterisk B, you still
 don't need to listen on two ports on Asterisk C.  Just leave off the port
 number in iax.conf the specification for Asterisk C on box B.  Likewise on
 Asterisk C, define the specification for Asterisk B to use port=4570.

 There is no need for port numbers to be the same on both machines when talking
 IAX.

   

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] Visual Dial Plan

2008-07-27 Thread randulo
On Sun, Jul 27, 2008 at 12:19 PM, Matt Watson [EMAIL PROTECTED] wrote:
 I've seen it before infact there is a website setup where people can
 post stuff made with it... kind of super nerdy!
 http://www.ratemydialplan.com

Cannot find lib path

too nerdy!

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] HASH, HASHKEYS, ClearHash explanation

2008-07-27 Thread Tilghman Lesher
On Sunday 27 July 2008 13:28:00 Pavel Jezek wrote:
 Hi, can somebody explain how to use this func/apps in asterisk?
 I tried to find some examples on mailinglists or wiki, however without
 success. thanks

The primary intended use is in conjunction with func_odbc, to allow you to
retrieve multiple values and reference them without having to assign each
field to an individual variable.  Think 'SELECT *' where the fields included
might change over time, and you don't want the placement of a minor field
to completely break your dialplan.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


Re: [asterisk-users] Visual Dial Plan

2008-07-27 Thread Paul Hales
randulo wrote:
 On Sun, Jul 27, 2008 at 12:19 PM, Matt Watson [EMAIL PROTECTED] wrote:
   
 I've seen it before infact there is a website setup where people can
 post stuff made with it... kind of super nerdy!
 http://www.ratemydialplan.com
 

 Cannot find lib path

 too nerdy!

   
Agreed - no lib, no site!

PaulH


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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


[asterisk-users] vmail.cgi and users.conf

2008-07-27 Thread George Pajari
Has anyone modified the voicemail web interface script (vmail.cgi) to 
work with users.conf

Currently vmail.cgi only looks in voicemail.conf for information on 
voicemail mailboxes but the new AsteriskGUI approach puts that 
information in users.conf.

-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
  www.netvoice.ca  www.ip-centrex.ca  www.ip-pbx.ca  www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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