date:20090213


This discussion is not making any sense to me.

Just what type of access product is this?

If you have fiber to the premise and are handed Ethernet from there to a
Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
Interconnection) type product.  It could also be framed over mid-band gear
over copper at some point in the circuit design and they could be fibbing
you on the fiber to the premise bit;  the fiber involved may actually be
a remote terminal or mux somewhere in the vicinity.  Either way, if you
have media converter CPE on your premises, this is an Ethernet product.

If that's so, there's no 512 kbps line.  There is no xDSL.  And there is
no incentive whatsoever to sell copper circuits as Ethernet transport is
usually more expensive and high-margin product.

Do you have a routed IP interface on your side?  If so, what equipment is
it on?  It's not the switch, as the switch is Layer 2.


On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen benny+use...@amorsen.dk
wrote:
 Vikas topg...@gmail.com writes:
 
 The ISP said that they ran a fiber optic wire to a media box at our
 office and from there there is a RJ45 to the switch. They bring no new
 equipment to our premises each time we provison a new port. Hence this
 upload speed limitation is not due to the copper wire.
 
 So the ISP is being deliberately difficult. I am assuming that their
 motivation is that they want to sell E1's instead of the 512kbps
 lines.
 
 You can fight your ISP by installing various multiplexing equipment,
 but it's an arms race, and they will probably win it -- losing you as
 a customer obviously doesn't worry them, while you're apparently
 willing to go to great lengths to stay with them.
 
 I would recomment just switching to E1 (preferably with a different
 provider). It's that or moving HQ to somewhere sane.
 
 
 /Benny
 
 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-users] PRI Test Lab

Hey Everyone,

I would like to start testing/playing with PRI channels but I don't have access 
to a PRI line.  Is it possible to do the equivilent of a crossover between two 
PRI Cards (say Digium's TE120P)?

What I was thinking is that I could set one asterisk box up with a PRI card set 
as the TE and provide clocking and another box exactly the same but with the 
card setup as NT.

I think I would also need to wire up the correct type of crossover as a 
standard ethernet crossover would not work or would it?

Thanks in advance.

Lee


  


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[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

2009-02-13 Thread joekane

Default FreePBX context,

[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include = ext-did-catchall; THIS MUST COME AFTER ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)

The call seems to be going here

[ext-did-catchall]
include = ext-did-catchall-custom
exten = s,1,Noop(No DID or CID Match)
exten = s,n(a2),Answer
exten = s,n,Wait(2)
exten = s,n,Playback(ss-noservice)
exten = s,n,SayAlpha(${FROM_DID})
exten = s,n,Hangup
exten = _.,1,Set(__FROM_DID=${EXTEN})
exten = _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten = _.,n,Goto(s,a2)
exten = h,1,Hangup

; end of [ext-did-catchall]

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Re: [asterisk-users] PRI Test Lab



On Fri, 13 Feb 2009 09:18:49 + (GMT), Lee Wilson leef...@yahoo.co.uk
wrote:
 Hey Everyone,
 
 I would like to start testing/playing with PRI channels but I don't have
 access to a PRI line.  Is it possible to do the equivilent of a crossover
 between two PRI Cards (say Digium's TE120P)?
 
 What I was thinking is that I could set one asterisk box up with a PRI
 card set as the TE and provide clocking and another box exactly the same
 but with the card setup as NT.
 
 I think I would also need to wire up the correct type of crossover as a
 standard ethernet crossover would not work or would it?

Standard Ethernet crossover will not work.  However, you can make a T1
crossover cable out of CAT-5 using a different pinout:

http://www.ldfacts.com/faq_files/How-to-Make-a-T1-Crossover-cable.htm

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

Oh--you mentioned in an earlier post that the Cisco switch was installed by
the ISP, so presumably that is something they consider their CPE as well.

You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950
does not have a Layer 3 feature set;  that only comes with MSFCs on
higher-order Catalysts.  So, they are doing in some fashion other than on
the switch ports, which is why I asked about the routed interfaces;  does
anything plugged into a given port have a separate routed interface?

-- Alex


On Fri, 13 Feb 2009 04:17:37 -0500, Alex Balashov
abalas...@evaristesys.com wrote:
 
 This discussion is not making any sense to me.
 
 Just what type of access product is this?
 
 If you have fiber to the premise and are handed Ethernet from there to a
 Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
 Interconnection) type product.  It could also be framed over mid-band
gear
 over copper at some point in the circuit design and they could be fibbing
 you on the fiber to the premise bit;  the fiber involved may actually
be
 a remote terminal or mux somewhere in the vicinity.  Either way, if you
 have media converter CPE on your premises, this is an Ethernet product.
 
 If that's so, there's no 512 kbps line.  There is no xDSL.  And there
is
 no incentive whatsoever to sell copper circuits as Ethernet transport is
 usually more expensive and high-margin product.
 
 Do you have a routed IP interface on your side?  If so, what equipment is
 it on?  It's not the switch, as the switch is Layer 2.
 
 
 On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen
 benny+use...@amorsen.dk
 wrote:
 Vikas topg...@gmail.com writes:

 The ISP said that they ran a fiber optic wire to a media box at our
 office and from there there is a RJ45 to the switch. They bring no new
 equipment to our premises each time we provison a new port. Hence this
 upload speed limitation is not due to the copper wire.

 So the ISP is being deliberately difficult. I am assuming that their
 motivation is that they want to sell E1's instead of the 512kbps
 lines.

 You can fight your ISP by installing various multiplexing equipment,
 but it's an arms race, and they will probably win it -- losing you as
 a customer obviously doesn't worry them, while you're apparently
 willing to go to great lengths to stay with them.

 I would recomment just switching to E1 (preferably with a different
 provider). It's that or moving HQ to somewhere sane.


 /Benny


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 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775
 
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Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-users] Cisco IP Phone 7940G.

Hello I recently get a Cisco 7940G IP Phone and I try to make several
things with it and I en counted many difficulties:

1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp server... I
downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
directory.
I don't get any luck here either. I look in the /var/log/messages and
I observed that my phone request 4 different files that i don't have
it in my tftp directory.
Here's my tftp output session with my phone:

Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
192.168.1.3:52178
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
to 192.168.1.3:52180
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
to 192.168.1.3:52181
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
192.168.1.3:52182

as you see my phone request 4 files that doesn't comes in archive
P0S3-08-11-00.zip:
SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
SEPDefault.cnf...

while my archive contents is the following:
OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
P0S3-08-11-00.sb2

3.) I want to make this phone to be SIP compatible. A friend of main
gave me a .cnf file with an example of configuration for SIP.
How may I rename this cnf file to make work with my phone.

4.) On the other side my phone doesn't have ringtone either. Any clue
how may I put ringtones on it?

I know is a lot of questions for you guys, but I browse on cisco.com
web site and google for hours and I don't get it any clue to make work
this phone in any way.

Thank you for help.

Jonson.

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Re: [asterisk-users] PRI Test Lab

  I would like to start testing/playing with PRI
 channels but I don't have
  access to a PRI line.  Is it possible to do the
 equivilent of a crossover
  between two PRI Cards (say Digium's TE120P)?
  
  What I was thinking is that I could set one asterisk
 box up with a PRI
  card set as the TE and provide clocking and another
 box exactly the same
  but with the card setup as NT.
  
  I think I would also need to wire up the correct type
 of crossover as a
  standard ethernet crossover would not work or would
 it?
 
 Standard Ethernet crossover will not work.  However, you
 can make a T1
 crossover cable out of CAT-5 using a different pinout:
 
 http://www.ldfacts.com/faq_files/How-to-Make-a-T1-Crossover-cable.htm
 

Alex, thanks for the quick response.

So I can assume from your response this should work.  That was easy :-)

I just want to clarify before I got and buy anything the cards are not so cheap.

Thanks again

Lee


  


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Re: [asterisk-users] Cisco IP Phone 7940G.


This phone is currently running the SCCP (Skinny) image.  Before you will
get anywhere you need to load the SIP firmware image onto it.  The SEP*
configuration files are for SCCP.

After doing that, the phone will start requesting the correct files.  You
may need to upgrade through various SIP images cumulatively.

On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. jonsonpla...@gmail.com
wrote:
 Hello I recently get a Cisco 7940G IP Phone and I try to make several
 things with it and I en counted many difficulties:
 
 1.) I tried to unlock the phone and to set manually IP Address,
 Netmask, Gateway etc. I don't get any luck.
 2.) I tried to upgrade firmware like they said with tftp server... I
 downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
 directory.
 I don't get any luck here either. I look in the /var/log/messages and
 I observed that my phone request 4 different files that i don't have
 it in my tftp directory.
 Here's my tftp output session with my phone:
 
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
 192.168.1.3:52178
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
 to 192.168.1.3:52180
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
 to 192.168.1.3:52181
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
 192.168.1.3:52182
 
 as you see my phone request 4 files that doesn't comes in archive
 P0S3-08-11-00.zip:
 SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
 SEPDefault.cnf...
 
 while my archive contents is the following:
 OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
 P0S3-08-11-00.sb2
 
 3.) I want to make this phone to be SIP compatible. A friend of main
 gave me a .cnf file with an example of configuration for SIP.
 How may I rename this cnf file to make work with my phone.
 
 4.) On the other side my phone doesn't have ringtone either. Any clue
 how may I put ringtones on it?
 
 I know is a lot of questions for you guys, but I browse on cisco.com
 web site and google for hours and I don't get it any clue to make work
 this phone in any way.
 
 Thank you for help.
 
 Jonson.
 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] PRI Test Lab



On Fri, 13 Feb 2009 09:48:11 + (GMT), Lee Wilson leef...@yahoo.co.uk
wrote:

 Alex, thanks for the quick response.
 
 So I can assume from your response this should work.  That was easy :-)
 
 I just want to clarify before I got and buy anything the cards are not so
 cheap.

Yep, it should work.  

I am not exactly sure how one goes about setting one of the cards to
provide the T1 master clock as I have only configured low-level T1 settings
on Sangoma cards, but it should be possible.  Although, honestly, I am not
sure that you're going to get a lot of timing slips on something like a 6
ft crossover cable anyway even if the clocks are both set to line or
internal;  timing synchronisation is much more of a concern on lengthier
spans and circuit designs that go through numerous network elements.

 Thanks again

Sure thing!

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status

2009-02-13 Thread Rayed Bs

you can get the originate resonse with a function dump_event and the
$asm-add_event_handler;off corse you can do it with a script php.

2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies) 
lloyd.aloys...@sunteltech.ca

 Dear All,

 I am originating the call directly to the SIP Provider using the maganger
 interface + originate (ASYNC)  command. Here is the PHP-AGI Script.

 $call = $asm-send_request('Originate',
  array('Channel'=SIP/416...@abc/n,
 'Context'='ORIG',
 'Exten'='s',
* 'Async'='1',*
 'MaxRetries' = '1',
 'RetryTime' = '10',
 'Priority'=1,
 'Account'=$phonenumber,
 'Callerid'=$callid)


 *extensions.conf*

 [ORIG]
 exten = s,1,Answer
 exten = s,2,Playback(ivrfile)
 exten = s,n,Hangup


 How Can I get the Originate Status using Async?  ANSWER, BUSY, NOANSWER
 etc..


 Thank you.
 Lloyd





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Re: [asterisk-users] Cisco IP Phone 7940G.

I understand, but i cannot load the new firmware... is any well know method?


On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
abalas...@evaristesys.com wrote:

 This phone is currently running the SCCP (Skinny) image.  Before you will
 get anywhere you need to load the SIP firmware image onto it.  The SEP*
 configuration files are for SCCP.

 After doing that, the phone will start requesting the correct files.  You
 may need to upgrade through various SIP images cumulatively.

 On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. jonsonpla...@gmail.com
 wrote:
 Hello I recently get a Cisco 7940G IP Phone and I try to make several
 things with it and I en counted many difficulties:

 1.) I tried to unlock the phone and to set manually IP Address,
 Netmask, Gateway etc. I don't get any luck.
 2.) I tried to upgrade firmware like they said with tftp server... I
 downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
 directory.
 I don't get any luck here either. I look in the /var/log/messages and
 I observed that my phone request 4 different files that i don't have
 it in my tftp directory.
 Here's my tftp output session with my phone:

 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
 192.168.1.3:52178
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
 to 192.168.1.3:52180
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
 to 192.168.1.3:52181
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
 192.168.1.3:52182

 as you see my phone request 4 files that doesn't comes in archive
 P0S3-08-11-00.zip:
 SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
 SEPDefault.cnf...

 while my archive contents is the following:
 OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
 P0S3-08-11-00.sb2

 3.) I want to make this phone to be SIP compatible. A friend of main
 gave me a .cnf file with an example of configuration for SIP.
 How may I rename this cnf file to make work with my phone.

 4.) On the other side my phone doesn't have ringtone either. Any clue
 how may I put ringtones on it?

 I know is a lot of questions for you guys, but I browse on cisco.com
 web site and google for hours and I don't get it any clue to make work
 this phone in any way.

 Thank you for help.

 Jonson.

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 asterisk-users mailing list
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 --
 Alex Balashov
 Evariste Systems
 Web    : http://www.evaristesys.com/
 Tel    : (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Cisco IP Phone 7940G.


Have a look at:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml#topic2

On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S. jonsonpla...@gmail.com
wrote:
 I understand, but i cannot load the new firmware... is any well know
 method?
 
 
 On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
 abalas...@evaristesys.com wrote:

 This phone is currently running the SCCP (Skinny) image.  Before you
 will
 get anywhere you need to load the SIP firmware image onto it.  The SEP*
 configuration files are for SCCP.

 After doing that, the phone will start requesting the correct files.
  You
 may need to upgrade through various SIP images cumulatively.

 On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
 jonsonpla...@gmail.com
 wrote:
 Hello I recently get a Cisco 7940G IP Phone and I try to make several
 things with it and I en counted many difficulties:

 1.) I tried to unlock the phone and to set manually IP Address,
 Netmask, Gateway etc. I don't get any luck.
 2.) I tried to upgrade firmware like they said with tftp server... I
 downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
 directory.
 I don't get any luck here either. I look in the /var/log/messages and
 I observed that my phone request 4 different files that i don't have
 it in my tftp directory.
 Here's my tftp output session with my phone:

 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
 192.168.1.3:52178
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
 to 192.168.1.3:52180
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
 to 192.168.1.3:52181
 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
 192.168.1.3:52182

 as you see my phone request 4 files that doesn't comes in archive
 P0S3-08-11-00.zip:
 SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
 SEPDefault.cnf...

 while my archive contents is the following:
 OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
 P0S3-08-11-00.sb2

 3.) I want to make this phone to be SIP compatible. A friend of main
 gave me a .cnf file with an example of configuration for SIP.
 How may I rename this cnf file to make work with my phone.

 4.) On the other side my phone doesn't have ringtone either. Any clue
 how may I put ringtones on it?

 I know is a lot of questions for you guys, but I browse on cisco.com
 web site and google for hours and I don't get it any clue to make work
 this phone in any way.

 Thank you for help.

 Jonson.

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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
    http://lists.digium.com/mailman/listinfo/asterisk-users
 --
 Alex Balashov
 Evariste Systems
 Web    : http://www.evaristesys.com/
 Tel    : (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread k4rjj








 

I'm trying to do the same and have read the mentioned sites. The one item I can't seem to get past is a working TFTP server. What is the easiest method to get one running and what packages in Linux or Windows work best?Thanks for putting up with a Linux newbie. Ronny

 -- Original message from Alex Balashov abalas...@evaristesys.com: --


 
 Have a look at:
 
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080
 094584.shtml#topic2
 
 On Fri, 13 Feb 2009 12:06:48 +0200, "Catalin S." 
 wrote:
  I understand, but i cannot load the new firmware... is any well know
  method?
  
  
  On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
   wrote:
 
  This phone is currently running the SCCP (Skinny) image. Before you
  will
  get anywhere you need to load the SIP firmware image onto it. The SEP*
  configuration files are for SCCP.
 
  After doing that, the phone will start requesting the correct files.
  You
  may need to upgrade through various SIP images cumulatively.
 
  On Fri, 13 Feb 2009 11:42:03 +0200, "Catalin S."
  
  wrote:
  Hello I recently get a Cisco 7940G IP Phone and I try to make several
  things with it and I en counted many difficulties:
 
  1.) I tried to unlock the phone and to set manually IP Address,
  Netmask, Gateway etc. I don't get any luck.
  2.) I tried to upgrade firmware like they said with tftp server... I
  downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
  directory.
  I don't get any luck here either. I look in the /var/log/messages and
  I observed that my phone request 4 different files that i don't have
  it in my tftp directory.
  Here's my tftp output session with my phone:
 
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
  192.168.1.3:52178
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
  SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
  to 192.168.1.3:52180
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
  to 192.168.1.3:52181
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
  192.168.1.3:52182
 
  as you see my phone request 4 files that doesn't comes in archive
  P0S3-08-11-00.zip:
  SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
  SEPDefault.cnf...
 
  while my archive contents is the following:
  OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
  P0S3-08-11-00.sb2
 
  3.) I want to make this phone to be SIP compatible. A friend of main
  gave me a .cnf file with an example of configuration for SIP.
  How may I rename this cnf file to make work with my phone.
 
  4.) On the other side my phone doesn't have ringtone either. Any clue
  how may I put ringtones on it?
 
  I know is a lot of questions for you guys, but I browse on cisco.com
  web site and google for hours and I don't get it any clue to make work
  this phone in any way.
 
  Thank you for help.
 
  Jonson.
 
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  Web  : http://www.evaristesys.com/
  Tel  : (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (678) 237-1775
 
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 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775
 
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Re: [asterisk-users] PRI Test Lab

Hey Gordon,

I was probably going with the Openvox anyway because they were cheaper. 

As this is just a hobby, no live environment I wanted to move away from just 
playing with SIP/IAX and get proper channels setup.

I was inspired by any earlier test I did at my last company where I was able to 
get a Cisco BRI Router working though the Asterisk PBX using BRIStuff. When I 
get it going, I'll try to document the setup in case anyone is interested.

Cheers

Lee


--- On Fri, 13/2/09, Gordon Henderson gor...@drogon.net wrote:

 From: Gordon Henderson gor...@drogon.net
 Subject: Re: [asterisk-users] PRI Test Lab
 To: Lee Wilson leef...@yahoo.co.uk
 Date: Friday, 13 February, 2009, 10:18 AM
 On Fri, 13 Feb 2009, Lee Wilson wrote:
 
  I just want to clarify before I got and buy anything
 the cards are not so cheap.
 
 Hi Lee,
 
 I'm assuming you're in the UK from your yahoo
 address (I'm in the UK too)
 
 I've installed a few boxes with these cards:
 
  
 http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html
 
 Not had any real issues with them and they're a bit
 cheaper than the digium cards.
 
 (I've never used one as a clock master though - did
 consider doing what you're doing before my first ISDN30
 install, but decided that as enough people were using them
 and saying that they worked, that it should just
 work and it did).
 
 Regards,
 
 Gordon
 -- www.drogon.net


  


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Re: [asterisk-users] PRI Test Lab

On 2/13/09, Alex Balashov abalas...@evaristesys.com wrote:


  On Fri, 13 Feb 2009 09:48:11 + (GMT), Lee Wilson leef...@yahoo.co.uk
  wrote:


   Alex, thanks for the quick response.
  
   So I can assume from your response this should work.  That was easy :-)
  
   I just want to clarify before I got and buy anything the cards are not so
   cheap.


 Yep, it should work.

  I am not exactly sure how one goes about setting one of the cards to
  provide the T1 master clock as I have only configured low-level T1 settings
  on Sangoma cards, but it should be possible.  Although, honestly, I am not
  sure that you're going to get a lot of timing slips on something like a 6
  ft crossover cable anyway even if the clocks are both set to line or
  internal;  timing synchronisation is much more of a concern on lengthier
  spans and circuit designs that go through numerous network elements.

If you want to be able to set Master timing you will have to use
Sangoma cards, they allow you force the timing in the wanrouter
configuration. I have done extensive testing with crossover T1/E1/PRIs
and I even have a testing lab set up with servers that have quad T1
cards with crossover cables going between all of the ports. There are
no time slip issues, and you can run the T1 line for hundreds of feet
without issues on good cable. For the cable I would suggest buying one
of these:

http://www.smartronixstore.com/index.cfm?fuseaction=product.displayProduct_ID=9
  (SuperLooper ISDN (PRI) Crossover Adapter)

They are fairly cheap and will save you the headache of making your own cables.

If you are using Sangoma cards you will have to configure Wanrouter
properly, but the Asterisk/zaptel side is fairly easy to configure,
just make sure one side is pri_net and the other is pri_cpe and you
should be good to go.

MATT---

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Re: [asterisk-users] PRI Test Lab

Matt Florell wrote:

 just make sure one side is pri_net and the other is pri_cpe and you
 should be good to go.

Ah, yes.  Definitely don't forget that one.  That trips a lot of folks up.

One side must be in ISDN user emulation mode and the other network.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] PRI Test Lab

I thought that, with PRI, it was possible to play using a single E1/T1 port
and a T1/E1 loopback adapter (a RJ45 plug with properly assigned wires which
makes any outgoing call an incoming one on another PRI channel) ?

For lab testing, this kind of trick seems very useful.
I've never tried it yet but is it correct ?

Cheers
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Re: [asterisk-users] PRI Test Lab

Olivier wrote:

 I thought that, with PRI, it was possible to play using a single E1/T1 
 port and a T1/E1 loopback adapter (a RJ45 plug with properly assigned 
 wires which makes any outgoing call an incoming one on another PRI 
 channel) ?
 
 For lab testing, this kind of trick seems very useful.
 I've never tried it yet but is it correct ?

No, because HDLC and ISDN Q.921 would not come up.  That requires two 
distinct endpoints and a bidirectional traffic exchange.

I think some Sangoma cards have a mode where you can put the port in a 
certain kind of loopback mode for this type of thing, but I am not 
sure--and in any case, that is a different concept.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] PRI Test Lab

2009-02-13 Thread Tom Moore

The cable needed for this is a different cable than an ethernet cross over.
I have actually done this same thing today with a Samsung 100 system and
Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great.

A question of my own:
I know I can emulate the network side of a pri connection, but can I do this
same trick with other t1 standards like ani and others?
If I can be a client on the different t1 types, does this also mean I can be
the server side and feed back the different standards to legacy equipment as
well or are there some limitations to this?
 
Thanks,
Tom

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Wilson
Sent: Friday, February 13, 2009 4:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI Test Lab

Hey Everyone,

I would like to start testing/playing with PRI channels but I don't have
access to a PRI line.  Is it possible to do the equivilent of a crossover
between two PRI Cards (say Digium's TE120P)?

What I was thinking is that I could set one asterisk box up with a PRI card
set as the TE and provide clocking and another box exactly the same but with
the card setup as NT.

I think I would also need to wire up the correct type of crossover as a
standard ethernet crossover would not work or would it?

Thanks in advance.

Lee


  


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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread D Tucny

A 2950 can be configured to limit the speed per port...

I guess the ISP here is operating this way because they are out of the way
and have limited bandwidth themselves, so, they are trying to split up the
bandwidth provided into smaller, more manageable chunks to avoid overloading
things at their end...

In asia here too the ISP that has service in this building has put in 24port
switches, if I ask for ethernet service, I'm told there's no such thing, all
I can order is ADSL, if I order ADSL, I get a 10Mb/s ethernet connection to
the switch, but then internet access is provided over PPPoE limited to 3Mb/s
both ways, I can get additional connections, to the same switches, with
seperate PPPoE accounts, again limited to 3Mb/s... So, at least I'm luckier
than Vikas, but, there is no alternative... There are features I would like,
but, in a monopoly you get what your given...

It's possible to load balance traffic over 4 connections though without any
help from the ISP... It won't be perfectly balanced, but it will do a
reasonably decent job... The options are many though and it depends on what
kit you have... I've done it with cisco routers before without nat where the
ISP was happy to support it and linux firewalls with nat with multiple
ISPs...

d

2009/2/13 Alex Balashov abalas...@evaristesys.com

 Oh--you mentioned in an earlier post that the Cisco switch was installed by
 the ISP, so presumably that is something they consider their CPE as well.

 You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950
 does not have a Layer 3 feature set;  that only comes with MSFCs on
 higher-order Catalysts.  So, they are doing in some fashion other than on
 the switch ports, which is why I asked about the routed interfaces;  does
 anything plugged into a given port have a separate routed interface?

 -- Alex


 On Fri, 13 Feb 2009 04:17:37 -0500, Alex Balashov
 abalas...@evaristesys.com wrote:
 
  This discussion is not making any sense to me.
 
  Just what type of access product is this?
 
  If you have fiber to the premise and are handed Ethernet from there to a
  Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
  Interconnection) type product.  It could also be framed over mid-band
 gear
  over copper at some point in the circuit design and they could be fibbing
  you on the fiber to the premise bit;  the fiber involved may actually
 be
  a remote terminal or mux somewhere in the vicinity.  Either way, if you
  have media converter CPE on your premises, this is an Ethernet product.
 
  If that's so, there's no 512 kbps line.  There is no xDSL.  And there
 is
  no incentive whatsoever to sell copper circuits as Ethernet transport is
  usually more expensive and high-margin product.
 
  Do you have a routed IP interface on your side?  If so, what equipment is
  it on?  It's not the switch, as the switch is Layer 2.
 
 
  On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen
  benny+use...@amorsen.dk benny%2buse...@amorsen.dk
  wrote:
  Vikas topg...@gmail.com writes:
 
  The ISP said that they ran a fiber optic wire to a media box at our
  office and from there there is a RJ45 to the switch. They bring no new
  equipment to our premises each time we provison a new port. Hence this
  upload speed limitation is not due to the copper wire.
 
  So the ISP is being deliberately difficult. I am assuming that their
  motivation is that they want to sell E1's instead of the 512kbps
  lines.
 
  You can fight your ISP by installing various multiplexing equipment,
  but it's an arms race, and they will probably win it -- losing you as
  a customer obviously doesn't worry them, while you're apparently
  willing to go to great lengths to stay with them.
 
  I would recomment just switching to E1 (preferably with a different
  provider). It's that or moving HQ to somewhere sane.
 
 
  /Benny
 
 
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  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (678) 237-1775
 
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 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Caller ID replacement

2009-02-13 Thread D Tucny

The hotdesking section of the asterisk book may also be of interest...

d

2009/2/13 David Ruggles da...@safedatausa.com

 Some googling lead me to this:
 http://hans.fugal.net/blog/tag/astdb

 Which looks like it has an answer.

 Thanks all!

 Thanks,

 David Ruggles
 CCNA MCSE (NT) CNA A+
 Network EngineerSafe Data, Inc.
 (910) 285-7200  da...@safedatausa.com



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
 Ruggles
 Sent: Thursday, February 12, 2009 12:24 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Caller ID replacement


 Could you give me an example of how this would look in the dialplan?

 Thanks,

 David Ruggles
 CCNA MCSE (NT) CNA A+
 Network EngineerSafe Data, Inc.
 (910) 285-7200  da...@safedatausa.com



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
 Nicholson
 Sent: Thursday, February 12, 2009 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Caller ID replacement


 On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote:
  I'm working on building a pbx that will allow us to use our cellphones as
  extensions (to some extent)
 
  The dialout is working fine. What I would like to do is have an inbound
  cellphone call appear as if it were an extension. So right now if I call
 in
  from cell #9995551212 the caller id is 9995551212 but if I dial extension
  30013 it will call cell #9995551212. I would like to change the caller id
 so
  9995551212 is changed to 30013 on the inbound call. Doing one is simple
  enough, but I would like have an easy (more or less) way of setting up
 some
  global variables that link the cell phone #'s and extensions and have
 this
  done somewhat automagically.

 I would implement this using the a database (astdb or odbc) containing
 the mapping from cell number to extension.  Then for each call that may
 need callerid modification, you can check the database for the proper
 mapping.  With this method it is also easy to add new mappings.

 --
 Matthew Nicholson
 Digium, Inc. | Software Developer


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Re: [asterisk-users] PRI Test Lab

On 2/13/09, Tom Moore tommym2...@gmail.com wrote:
 The cable needed for this is a different cable than an ethernet cross over.
  I have actually done this same thing today with a Samsung 100 system and
  Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great.

I would again just recommend getting one of these, they are worth it:
http://www.smartronixstore.com/index.cfm?fuseaction=product.displayProduct_ID=9
 (SuperLooper ISDN (PRI) Crossover Adapter)

  A question of my own:
  I know I can emulate the network side of a pri connection, but can I do this
  same trick with other t1 standards like ani and others?
  If I can be a client on the different t1 types, does this also mean I can be
  the server side and feed back the different standards to legacy equipment as
  well or are there some limitations to this?

Yes you can do other T1/E1 standards like RBS D4/AMI, EM wink start
and most of your other old favorites


MATT---

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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread Vikas

The ISP tells me that it is a Metro Ethernet product.

Here is a picture of the box made by Mc Mans tel where the ISP
inputs a fat black wire. I have never heard of Mc Mans tel and
googling it comes up with nothing.

http://www.grmtech.com/blog/wp-content/uploads/2009/02/img_0035_1-300x225.jpg

From this McManstel box the output is two yellow CAT5 cables. Here is
a picture showing the wiring on this McManstel box. Interesting thing
to note is that McManstel box does not even take a power supply so it
cannot be running its own processor or OS.

http://www.grmtech.com/blog/wp-content/uploads/2009/02/black-wire-input-and-two-eth-yellow-wire-output-300x225.jpg

These two yellow wires go to the CISCO Catalyst 2950 switch. Here is a
picture showing the input to the CISCO switch:

http://www.grmtech.com/blog/wp-content/uploads/2009/02/input-to-the-cisco-switch-300x128.jpg

From the CISCO switch I have two wires coming out and going to my two
different stand alone linux server which act as my routers. Here is a
picture showing the output from the CISCO switch going to the two
linux servers:

http://www.grmtech.com/blog/wp-content/uploads/2009/02/cisco2950-24ports-farleft-two-output-300x89.jpg

My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic cable ?
2. What is the Mc Manstel box doing ?
3. What CISCO router do I need to buy to do bandwidth aggregation at my end ?

I have made a blog post with pictures and the problem statement that I
will keep updated as I learn more about the problem and the eventual
solution. The link to the blog post is at:
http://www.grmtech.com/blog/kolkata-broadband/

If you need any infroamtion from me let me know and I will find it and
post it here. Some of the technicians working for the ISP have been
helpful so if there is some question that I can ask them to be able to
figure out what is going on here let me know and I will ask the
technicians from the ISP and post the responses here.

Thanks once again for taking time out to help me.

On Fri, Feb 13, 2009 at 3:30 AM, Alex Balashov
abalas...@evaristesys.com wrote:
Oh--you mentioned in an earlier post that the Cisco switch was installed by
the ISP, so presumably that is something they consider their CPE as well.

You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950
does not have a Layer 3 feature set; that only comes with MSFCs on
higher-order Catalysts. So, they are doing in some fashion other than on
the switch ports, which is why I asked about the routed interfaces; does
anything plugged into a given port have a separate routed interface?

-- Alex

On Fri, 13 Feb 2009 04:17:37 -0500, Alex Balashov
abalas...@evaristesys.com wrote:

This discussion is not making any sense to me.

Just what type of access product is this?

If you have fiber to the premise and are handed Ethernet from there to a
Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
Interconnection) type product. It could also be framed over mid-band
gear
over copper at some point in the circuit design and they could be fibbing
you on the fiber to the premise bit; the fiber involved may actually
be
a remote terminal or mux somewhere in the vicinity. Either way, if you
have media converter CPE on your premises, this is an Ethernet product.

If that's so, there's no 512 kbps line. There is no xDSL. And there
is
no incentive whatsoever to sell copper circuits as Ethernet transport is
usually more expensive and high-margin product.

Do you have a routed IP interface on your side? If so, what equipment is
it on? It's not the switch, as the switch is Layer 2.

On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen
benny+use...@amorsen.dk
wrote:
Vikas topg...@gmail.com writes:

The ISP said that they ran a fiber optic wire to a media box at our
office and from there there is a RJ45 to the switch. They bring no new
equipment to our premises each time we provison a new port. Hence this
upload speed limitation is not due to the copper wire.

So the ISP is being deliberately difficult. I am assuming that their
motivation is that they want to sell E1's instead of the 512kbps
lines.

You can fight your ISP by installing various multiplexing equipment,
but it's an arms race, and they will probably win it -- losing you as
a customer obviously doesn't worry them, while you're apparently
willing to go to great lengths to stay with them.

I would recomment just switching to E1 (preferably with a different
provider). It's that or moving HQ to somewhere sane.

/Benny

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Mobile :

Re: [asterisk-users] PRI Test Lab

2009/2/13 Alex Balashov abalas...@evaristesys.com

 Olivier wrote:

  I thought that, with PRI, it was possible to play using a single E1/T1
  port and a T1/E1 loopback adapter (a RJ45 plug with properly assigned
  wires which makes any outgoing call an incoming one on another PRI
  channel) ?
 
  For lab testing, this kind of trick seems very useful.
  I've never tried it yet but is it correct ?

 No, because HDLC and ISDN Q.921 would not come up.  That requires two
 distinct endpoints and a bidirectional traffic exchange.


Thanks  for correcting me !



 I think some Sangoma cards have a mode where you can put the port in a
 certain kind of loopback mode for this type of thing, but I am not
 sure--and in any case, that is a different concept.

 --
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 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Cisco IP Phone 7940G.

hey finally i did it. I upgraded the firmware to the latest sip
firmware and now i have the another problem. The requested files are
the following:

---///---
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf
to 192.168.1.3:51253
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to
192.168.1.3:51254
---///---

I made my own sip configuration in SIP00141CAA4B4C.cnf where
00141CAA4B4C is the mac address of phone, but i don't know what to
write in CTLSEP00141CAA4B4C.tlv, SEP00141CAA4B4C.cnf.xml and
SIPDefault.cnf. On display of the screen of phone all I have is Tftp
file missing... probably it expect all these files.

Anyway, Ronny I can give you my archive with what i had in my tftp and
i succeeded to update firmware. Just tell me if you want to send on
your personal e-mail these files.

Thank you guys for your help and interest.

On 2/13/09, k4...@bellsouth.net k4...@bellsouth.net wrote:



  I'm trying to do the same and have read the mentioned sites.  The one item
 I can't seem to get past is a working TFTP server.  What is the easiest
 method to get one running and what packages in Linux or Windows work best?

 Thanks for putting up with a Linux newbie.

 Ronny


  -- Original message from Alex Balashov
 abalas...@evaristesys.com: --


 
  Have a look at:
 
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080
  094584.shtml#topic2
 
  On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
  wrote:
   I understand, but i cannot load the new firmware... is any well know
   method?
  
  
   On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
   wrote:
  
   This phone is currently running the SCCP (Skinny) image.  Before you
   will
   get anywhere you need to load the SIP firmware image onto it.  The SEP*
   configuration files are for SCCP.
  
   After doing that, the phone will start requesting the correct files.
You
   may need to upgrade through various SIP images cumulatively.
  
   On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
  
   wrote:
   Hello I recently get a Cisco 7940G IP Phone and I try to make several
   things with it and I en counted many difficulties:
  
   1.) I tried to unlock the phone and to set manually IP Address,
   Netmask, Gateway etc. I don't get any luck.
   2.) I tried to upgrade firmware like they said with tftp server... I
   downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
   directory.
   I don't get any luck here either. I look in the /var/log/messages and
   I observed that my phone request 4 different files that i don't have
   it in my tftp directory.
   Here's my tftp output session with my phone:
  
   Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
   192.168.1.3:52178
   Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
   SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
   Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
   to 192.168.1.3:52180
   Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
   to 192.168.1.3:52181
   Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
   192.168.1.3:52182
  
   as you see my phone request 4 files that doesn't comes in archive
   P0S3-08-11-00.zip:
   SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
   SEPDefault.cnf...
  
   while my archive contents is the following:
   OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
   P0S3-08-11-00.sb2
  
   3.) I want to make this phone to be SIP compatible. A friend of main
   gave me a .cnf file with an example of configuration for SIP.
   How may I rename this cnf file to make work with my phone.
  
   4.) On the other side my phone doesn't have ringtone either. Any clue
   how may I put ringtones on it?
  
   I know is a lot of questions for you guys, but I browse on cisco.com
   web site and google for hours and I don't get it any clue to make work
   this phone in any way.
  
   Thank you for help.
  
   Jonson.
  
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   Tel: (+1) (678) 954-0670
   Direct : (+1) (678) 954-0671
   Mobile : (+1) (678) 237-1775
  
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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread pe...@networkoblivion.com

Windows - http://kin.klever.net/pumpkin/binaries

Works great.

k4...@bellsouth.net wrote:
 
  I'm trying to do the same and have read the mentioned sites.  The one 
 item I can't seem to get past is a working TFTP server.  What is the 
 easiest method to get one running and what packages in Linux or Windows 
 work best?
 
 Thanks for putting up with a Linux newbie. 
 
 Ronny
 
 -- Original message from Alex Balashov
 abalas...@evaristesys.com: --
 
 
  
   Have a look at:
  
 
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080
 
   094584.shtml#topic2
  
   On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
   wrote:
I understand, but i cannot load the new firmware... is any well
 know
method?
   
   
On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
wrote:
   
This phone is currently running the SCCP (Skinny) image.
  Before you
will
get anywhere you need to load the SIP firmware image onto it.
  The SEP*
configuration files are for SCCP.
   
After doing that, the phone will start requesting the correct
 files.
 You
may need to upgrade through various SIP images cumulatively.
   
On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
   
wrote:
Hello I recently get a Cisco 7940G IP Phone and I try to make
 several
things with it and I en counted many difficulties:
   
1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp
 server... I
downloaded: P0S3-08-11-00.zip and I uncompressed the files in
 tftpboot
directory.
I don't get any luck here either. I look in the
 /var/log/messages and
I observed that my phone request 4 different files that i
 don't have
it in my tftp directory.
Here's my tftp output session with my phone:
   
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
192.168.1.3:52178
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 XMLDefault.cnf.xml
to 192.168.1.3:52180
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf
to 192.168.1.3:52181
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEPDefault.cnf to
192.168.1.3:52182
   
as you see my phone request 4 files that doesn't comes in
 archive
P0S3-08-11-00.zip:
SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml,
 SEP00141CAA4B4C.cnf,
SEPDefault.cnf...
   
while my archive contents is the following:
OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn,
 P0S3-08-11-00.loads,
P0S3-08-11-00.sb2
   
3.) I want to make this phone to be SIP compatible. A friend
 of main
gave me a .cnf file with an example of configuration for SIP.
How may I rename this cnf file to make work with my phone.
   
4.) On the other side my phone doesn't have ringtone either.
 Any clue
how may I put ringtones on it?
   
I know is a lot of questions for you guys, but I browse on
 cisco.com
web site and google for hours and I don't get it any clue to
 make work
this phone in any way.
   
Thank you for help.
   
Jonson.
   
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Evariste Systems
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Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
   
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   Tel : (+1) (678) 954-0670
   Direct : (+1) (678) 954-0671
   Mobile : (+1) (678) 237-1775

Re: [asterisk-users] PRI Test Lab

Matt Florell wrote:

 If you want to be able to set Master timing you will have to use
 Sangoma cards, they allow you force the timing in the wanrouter
 configuration. I have done extensive testing with crossover T1/E1/PRIs

I don't believe this is true; we use Digium cards connected back-to-back
to each other, with one providing the span timing (clock) all the time.
This is a very common configuration and works fine.

To set a Zaptel/DAHDI card to provide span timing, just set all the
spans on it to 'zero' as the priority for timing source; when none of
the span clocks are the timing source, the onboard clock on the card
will be the timing source.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] WiFi SIP phone w/VPN?

2009/2/12 Ken D'Ambrosio k...@jots.org

 Hi, all.  My subject line says it all: is there a WiFi SIP phone with VPN
 abilities?  Failing that, a WiFi phone that runs Linux?  I already know
 one phone that does meet my requirements -- the iPhone.  The new software
 comes with a Cisco VPN client, and a SIP client can be had from
 third-party vendors for jailbroken phones.


Could you elaborate this latest point (SIP clients avalability) ?

When I looked at it, iPhone developper agreement wouldn't allow third party
application to run in background, which is a show-stopper for getting
incoming calls.
Rumours say it will be changing with firmware 3.0 ...




  And, while I'm not averse to
 the idea,
 a) it ain't cheap, and
 b) it's a bit hack.

 I've googled my heart out, but haven't found anything else that (I'm sure)
 meets all three requirements.

 Thanks!

 -Ken


 --
 This message has been scanned for viruses and
 dangerous content by MailScanner, and is
 believed to be clean.


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Re: [asterisk-users] PRI Test Lab

2009/2/13 Kevin P. Fleming kpflem...@digium.com

 Matt Florell wrote:

  If you want to be able to set Master timing you will have to use
  Sangoma cards, they allow you force the timing in the wanrouter
  configuration. I have done extensive testing with crossover T1/E1/PRIs

 I don't believe this is true; we use Digium cards connected back-to-back
 to each other, with one providing the span timing (clock) all the time.
 This is a very common configuration and works fine.

 To set a Zaptel/DAHDI card to provide span timing, just set all the
 spans on it to 'zero' as the priority for timing source; when none of
 the span clocks are the timing source, the onboard clock on the card
 will be the timing source.


What if using a patch cord from port to another ?
Would you still value to zero each span timing priority or would you set one
end to 0 and the other to 1 ?



 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kpflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] PRI Test Lab

Olivier wrote:

 What if using a patch cord from port to another ?
 Would you still value to zero each span timing priority or would you set
 one end to 0 and the other to 1 ?

For connecting spans on a single card to each other, all the spans
should be set to '0'; there is no need to use a recovered clock for any
spans when they are all on the same card and using the same clock for
transmission.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am



On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote:


 Hello Asterisk Users and those with an Interest in VoIP Tech,


[snip]

Is there a Chicago area users group?  If not is there any interest in 
creating one?

j

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Re: [asterisk-users] PRI Test Lab

2009-02-13 Thread Dave Fullerton

Lee Wilson wrote:
 Hey Everyone,
 
 I would like to start testing/playing with PRI channels but I don't have 
 access to a PRI line.  Is it possible to do the equivilent of a crossover 
 between two PRI Cards (say Digium's TE120P)?
 
 What I was thinking is that I could set one asterisk box up with a PRI card 
 set as the TE and provide clocking and another box exactly the same but with 
 the card setup as NT.
 
 I think I would also need to wire up the correct type of crossover as a 
 standard ethernet crossover would not work or would it?
 
 Thanks in advance.
 
 Lee

Since you have gotten plenty of responses on this I thought I'd throw 
out another option. If you just want to play with how a PRI connection 
behaves without all the hardware investment, you can emulate a PRI over 
TDMOE. I did it a while back just to see how calls were passed back and 
forth and how result codes were set. Everything is configured the same 
in asterisk, you just use a dynamic span instead of a physical one. You 
will still need one side to have a timing source (I did get mine to work 
with just ztdummy).

-Dave

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Re: [asterisk-users] PRI Test Lab

Thanks Dave, I hadn't read up on Dynamic Spans before and will certainly take a 
look.  If anyone else is interested I did a quick google search and came up 
with the following links:

http://www.microalcarria.com/descargas/documentos/Linux/varios/Asterisk/asteriskdocs-docbook/docs-html/x1320.html

http://www.voip-info.org/wiki/view/Asterisk+TDMoE

If this is the correct stuff, it will give me something to start with.

Lee


--- On Fri, 13/2/09, Dave Fullerton dfullertaster...@shorelinecontainer.com 
wrote:

 From: Dave Fullerton dfullertaster...@shorelinecontainer.com
 Subject: Re: [asterisk-users] PRI Test Lab
 To: leef...@yahoo.co.uk, Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Friday, 13 February, 2009, 2:00 PM
 Lee Wilson wrote:
  Hey Everyone,
  
  I would like to start testing/playing with PRI
 channels but I don't have access to a PRI line.  Is it
 possible to do the equivilent of a crossover between two PRI
 Cards (say Digium's TE120P)?
  
  What I was thinking is that I could set one asterisk
 box up with a PRI card set as the TE and provide clocking
 and another box exactly the same but with the card setup as
 NT.
  
  I think I would also need to wire up the correct type
 of crossover as a standard ethernet crossover would not work
 or would it?
  
  Thanks in advance.
  
  Lee
 
 Since you have gotten plenty of responses on this I thought
 I'd throw out another option. If you just want to play
 with how a PRI connection behaves without all the hardware
 investment, you can emulate a PRI over TDMOE. I did it a
 while back just to see how calls were passed back and forth
 and how result codes were set. Everything is configured the
 same in asterisk, you just use a dynamic span instead of a
 physical one. You will still need one side to have a timing
 source (I did get mine to work with just ztdummy).
 
 -Dave


  


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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?



On Fri, 13 Feb 2009, Vikas wrote:

 In my opinion the only strategy that has a high probability of success is:

 Get a Cisco with five ethernet ports.  Use one for your connection to 
 asterisk.  Use the other four as your connection to the ISP, and MUX them.

 Can you please point me to some resource on how to MUX ?


[snip]

Actually that was a bit tongue-in-cheek as I didn't think it was a serious 
option due to cost.  Also I don't think you can properly MUX these lines 
without a similarly configured Cisco on the other side.

How about BigIP?  They make multi-upstream load balancer products.  Again 
very pricy, but I think it is exactly what you are looking for.

http://www.f5.com/solutions/availability/link-load-balancing/

j

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Re: [asterisk-users] After Monitor() files disappear

2009-02-13 Thread Gunnar Schaller

I'm moving over to asterisk-dev. Seems to be a bug.

Greetings,
Gunnar Schaller


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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread D Tucny

2009/2/13 Vikas topg...@gmail.com

 My questions are:
 1. The black wire coming into the Mc Manstel box is that a fibre optic
 cable ?
 2. What is the Mc Manstel box doing ?
 3. What CISCO router do I need to buy to do bandwidth aggregation at my end
 ?


1) Yes
2) It's stopping you from poking the fibre... the point where they spliced
the fibre coming in to the fibre running to the switch is on that plastic
box...
3) A cisco router with at least 5 ethernet interfaces, which would likely be
expensive, but, look at
http://www.cisco.com/en/US/tech/tk648/tk361/technologies_configuration_example09186a0080950834.shtmlfor
more info...

d
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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread Ronny Julian


Please send on!  Thanks!  What TFTP server did you use?


Catalin S. wrote:

hey finally i did it. I upgraded the firmware to the latest sip
firmware and now i have the another problem. The requested files are
the following:

---///---
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf
to 192.168.1.3:51253
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to
192.168.1.3:51254
---///---

I made my own sip configuration in SIP00141CAA4B4C.cnf where
00141CAA4B4C is the mac address of phone, but i don't know what to
write in CTLSEP00141CAA4B4C.tlv, SEP00141CAA4B4C.cnf.xml and
SIPDefault.cnf. On display of the screen of phone all I have is Tftp
file missing... probably it expect all these files.

Anyway, Ronny I can give you my archive with what i had in my tftp and
i succeeded to update firmware. Just tell me if you want to send on
your personal e-mail these files.

Thank you guys for your help and interest.

On 2/13/09, k4...@bellsouth.net k4...@bellsouth.net wrote:
  


 I'm trying to do the same and have read the mentioned sites.  The one item
I can't seem to get past is a working TFTP server.  What is the easiest
method to get one running and what packages in Linux or Windows work best?

Thanks for putting up with a Linux newbie.

Ronny


 -- Original message from Alex Balashov
abalas...@evaristesys.com: --




Have a look at:

  

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080


094584.shtml#topic2

On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
wrote:
  

I understand, but i cannot load the new firmware... is any well know
method?


On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
wrote:


This phone is currently running the SCCP (Skinny) image.  Before you
  

will


get anywhere you need to load the SIP firmware image onto it.  The SEP*
configuration files are for SCCP.

After doing that, the phone will start requesting the correct files.
  

 You


may need to upgrade through various SIP images cumulatively.

On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
  
wrote:
  

Hello I recently get a Cisco 7940G IP Phone and I try to make several
things with it and I en counted many difficulties:

1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp server... I
downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot
directory.
I don't get any luck here either. I look in the /var/log/messages and
I observed that my phone request 4 different files that i don't have
it in my tftp directory.
Here's my tftp output session with my phone:

Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
192.168.1.3:52178
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml
to 192.168.1.3:52180
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf
to 192.168.1.3:52181
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to
192.168.1.3:52182

as you see my phone request 4 files that doesn't comes in archive
P0S3-08-11-00.zip:
SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf,
SEPDefault.cnf...

while my archive contents is the following:
OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads,
P0S3-08-11-00.sb2

3.) I want to make this phone to be SIP compatible. A friend of main
gave me a .cnf file with an example of configuration for SIP.
How may I rename this cnf file to make work with my phone.

4.) On the other side my phone doesn't have ringtone either. Any clue
how may I put ringtones on it?

I know is a lot of questions for you guys, but I browse on cisco.com
web site and google for hours and I don't get it any clue to make work
this phone in any way.

Thank you for help.

Jonson.

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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread John Novack



k4...@bellsouth.net wrote:
  I'm trying to do the same and have read the mentioned sites.  The one 
 item I can't seem to get past is a working TFTP server.  What is the 
 easiest method to get one running and what packages in Linux or 
 Windows work best?

Solar winds TFTP server, free, is certainly the best, easy to install in 
a windows box and gives you an on screen log to view what has and has 
not been  successful

John Novack

-- 
Dog is my co-pilot


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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread John Novack



Catalin S. wrote:
 hey finally i did it. I upgraded the firmware to the latest sip firmware and 
 now i have the another problem. The requested files are the following:

 ---///---
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf
 to 192.168.1.3:51253
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to
 192.168.1.3:51254
 ---///---

 I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is 
 the mac address of phone, but i don't know what to write in 
 CTLSEP00141CAA4B4C.tlv,
Create an empty file and it will be happy. At least that has been my 
experience with my 7960.

Others can probably provide a sample of the remaining files.

So far I have been unable to go beyond version 7 firmware, as it is 
unhappy with the XML file when trying to move to version 8.

John Novack

-- 
Dog is my co-pilot


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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread Vikas

I have two more questions:

1. Is it a technical reason that the ISP has restricted the upload to
512 Kbps or is it a Marketing reason that they have restricted the
upload ?
2. Can I boot the cisco switch in run level 1 and modify the rate
limits on each of the ports ?
This vidoe talks about booting into run level 1:
http://www.youtube.com/watch?v=_VY5B6cTkT8

There are multiple tutorials to modify the rate limits on google search.

Obviously I am NOT going to do it without getting an ok from the ISP
but is this all that is needed ?

What would do if you found yourself in such a situation ?

Thanks once again for such valuable feedback.

On Fri, Feb 13, 2009 at 8:21 AM, D Tucny d...@tucny.com wrote:
2009/2/13 Vikas topg...@gmail.com

My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic
cable ?
2. What is the Mc Manstel box doing ?
3. What CISCO router do I need to buy to do bandwidth aggregation at my
end ?

1) Yes
2) It's stopping you from poking the fibre... the point where they spliced
the fibre coming in to the fibre running to the switch is on that plastic
box...
3) A cisco router with at least 5 ethernet interfaces, which would likely be
expensive, but, look at
http://www.cisco.com/en/US/tech/tk648/tk361/technologies_configuration_example09186a0080950834.shtml
for more info...

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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread Ronny Julian


Thanks!  I'll have to load it on my Wife's Windows machine.  I have gone 
total Linux and the darn things keep going!  My wife's Vista WinBlows 
laptop has to be booted every few days.

pe...@networkoblivion.com wrote:
 Windows - http://kin.klever.net/pumpkin/binaries

 Works great.

 k4...@bellsouth.net wrote:
   
  I'm trying to do the same and have read the mentioned sites.  The one 
 item I can't seem to get past is a working TFTP server.  What is the 
 easiest method to get one running and what packages in Linux or Windows 
 work best?

 Thanks for putting up with a Linux newbie. 

 Ronny

 -- Original message from Alex Balashov
 abalas...@evaristesys.com: --


  
   Have a look at:
  
 
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080

   094584.shtml#topic2
  
   On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
   wrote:
I understand, but i cannot load the new firmware... is any well
 know
method?
   
   
On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
wrote:
   
This phone is currently running the SCCP (Skinny) image.
  Before you
will
get anywhere you need to load the SIP firmware image onto it.
  The SEP*
configuration files are for SCCP.
   
After doing that, the phone will start requesting the correct
 files.
 You
may need to upgrade through various SIP images cumulatively.
   
On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
   
wrote:
Hello I recently get a Cisco 7940G IP Phone and I try to make
 several
things with it and I en counted many difficulties:
   
1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp
 server... I
downloaded: P0S3-08-11-00.zip and I uncompressed the files in
 tftpboot
directory.
I don't get any luck here either. I look in the
 /var/log/messages and
I observed that my phone request 4 different files that i
 don't have
it in my tftp directory.
Here's my tftp output session with my phone:
   
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
192.168.1.3:52178
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 XMLDefault.cnf.xml
to 192.168.1.3:52180
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf
to 192.168.1.3:52181
Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEPDefault.cnf to
192.168.1.3:52182
   
as you see my phone request 4 files that doesn't comes in
 archive
P0S3-08-11-00.zip:
SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml,
 SEP00141CAA4B4C.cnf,
SEPDefault.cnf...
   
while my archive contents is the following:
OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn,
 P0S3-08-11-00.loads,
P0S3-08-11-00.sb2
   
3.) I want to make this phone to be SIP compatible. A friend
 of main
gave me a .cnf file with an example of configuration for SIP.
How may I rename this cnf file to make work with my phone.
   
4.) On the other side my phone doesn't have ringtone either.
 Any clue
how may I put ringtones on it?
   
I know is a lot of questions for you guys, but I browse on
 cisco.com
web site and google for hours and I don't get it any clue to
 make work
this phone in any way.
   
Thank you for help.
   
Jonson.
   
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
   
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Re: [asterisk-users] Hangup extensions via CLI?

2009-02-13 Thread Tim Nelson

You guys think YOU'RE overdoing it... your solution works with a single line. 
My solution was some convoluted 100 line shell script! 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 

- Lenz Emilitri wrote: 
 

I have a feeling we're overdoing it :) 

l. 
 
 
2009/2/12 Lukas Rypl  r...@marconi.ttc.cz  
 



  asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep 
  SIP/7000 
 
 
 Hi, 
 
 I used this way of processing output from asterisk 1.2 and found out 
 that it is not 100% safe because there can appear unprintable characters 
 in the output. This will cause the following grep command to show 
 message similar to Binary content: matched instead of expected line. 
 
 It is necessary to use strings -a to filter output. So your example 
 should be: 
 
 asterisk -rx 'core show channels' | strings -a | grep SIP/7000 
 
 
 
 Hope it helps 
 
 Lukas 
 


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

 -- 
 Loway - home of QueueMetrics - http://queuemetrics.com 
 
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Re: [asterisk-users] Cisco IP Phone 7940G.


If you are running Linux you already have a TFTP server.  You just need to 
enable it.

j

On Fri, 13 Feb 2009, Ronny Julian wrote:


 Thanks!  I'll have to load it on my Wife's Windows machine.  I have gone
 total Linux and the darn things keep going!  My wife's Vista WinBlows
 laptop has to be booted every few days.

 pe...@networkoblivion.com wrote:
 Windows - http://kin.klever.net/pumpkin/binaries

 Works great.

 k4...@bellsouth.net wrote:

  I'm trying to do the same and have read the mentioned sites.  The one
 item I can't seem to get past is a working TFTP server.  What is the
 easiest method to get one running and what packages in Linux or Windows
 work best?

 Thanks for putting up with a Linux newbie.

 Ronny

 -- Original message from Alex Balashov
 abalas...@evaristesys.com: --


 
  Have a look at:
 
 
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080

  094584.shtml#topic2
 
  On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
  wrote:
  I understand, but i cannot load the new firmware... is any well
 know
  method?
 
 
  On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
  wrote:
 
  This phone is currently running the SCCP (Skinny) image.
  Before you
  will
  get anywhere you need to load the SIP firmware image onto it.
  The SEP*
  configuration files are for SCCP.
 
  After doing that, the phone will start requesting the correct
 files.
   You
  may need to upgrade through various SIP images cumulatively.
 
  On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
 
  wrote:
  Hello I recently get a Cisco 7940G IP Phone and I try to make
 several
  things with it and I en counted many difficulties:
 
  1.) I tried to unlock the phone and to set manually IP Address,
  Netmask, Gateway etc. I don't get any luck.
  2.) I tried to upgrade firmware like they said with tftp
 server... I
  downloaded: P0S3-08-11-00.zip and I uncompressed the files in
 tftpboot
  directory.
  I don't get any luck here either. I look in the
 /var/log/messages and
  I observed that my phone request 4 different files that i
 don't have
  it in my tftp directory.
  Here's my tftp output session with my phone:
 
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
  192.168.1.3:52178
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
  SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 XMLDefault.cnf.xml
  to 192.168.1.3:52180
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf
  to 192.168.1.3:52181
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEPDefault.cnf to
  192.168.1.3:52182
 
  as you see my phone request 4 files that doesn't comes in
 archive
  P0S3-08-11-00.zip:
  SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml,
 SEP00141CAA4B4C.cnf,
  SEPDefault.cnf...
 
  while my archive contents is the following:
  OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn,
 P0S3-08-11-00.loads,
  P0S3-08-11-00.sb2
 
  3.) I want to make this phone to be SIP compatible. A friend
 of main
  gave me a .cnf file with an example of configuration for SIP.
  How may I rename this cnf file to make work with my phone.
 
  4.) On the other side my phone doesn't have ringtone either.
 Any clue
  how may I put ringtones on it?
 
  I know is a lot of questions for you guys, but I browse on
 cisco.com
  web site and google for hours and I don't get it any clue to
 make work
  this phone in any way.
 
  Thank you for help.
 
  Jonson.
 
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  --
  Alex Balashov
  Evariste Systems
  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (678) 237-1775
 
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  Evariste Systems

Re: [asterisk-users] PRI Test Lab

On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
 Matt Florell wrote:

   If you want to be able to set Master timing you will have to use
   Sangoma cards, they allow you force the timing in the wanrouter
   configuration. I have done extensive testing with crossover T1/E1/PRIs


 I don't believe this is true; we use Digium cards connected back-to-back
  to each other, with one providing the span timing (clock) all the time.
  This is a very common configuration and works fine.

  To set a Zaptel/DAHDI card to provide span timing, just set all the
  spans on it to 'zero' as the priority for timing source; when none of
  the span clocks are the timing source, the onboard clock on the card
  will be the timing source.

Can you tell me where the setting is to force Master timing on Digium
cards per port? I really didn't think Digium cards had the ability to
force Master in this way. I've tried to do it with channelbanks before
and couldn't force it to master, whereas I can get it to work with
Sangoma.

Thanks,

MATT---

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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread David Backeberg

On Fri, Feb 13, 2009 at 9:58 AM, Vikas topg...@gmail.com wrote:
 What would do if you found yourself in such a situation ?

I would switch to a phone company phone line. An E1 or T1.

It sounds like this company is a data provider and not a phone company.

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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread John Novack

The solar winds windows server is very nice while troubleshooting 
though, as you have an on screen log that gives real time reports if 
something is in error.
Once all is working, have the Linux TFTP available for the phone when it 
needs it.
And sometimes enabling and KEEPING it enabled can be a challenge in itself.
Besides, he did say he was a noobie!


Can't speak to Vista , the solar winds works well on Win2K

John Novack


Jeff LaCoursiere wrote:
 If you are running Linux you already have a TFTP server.  You just need to 
 enable it.

 j

 On Fri, 13 Feb 2009, Ronny Julian wrote:

   
 Thanks!  I'll have to load it on my Wife's Windows machine.  I have gone
 total Linux and the darn things keep going!  My wife's Vista WinBlows
 laptop has to be booted every few days.

 pe...@networkoblivion.com wrote:
 
 Windows - http://kin.klever.net/pumpkin/binaries

 Works great.

 k4...@bellsouth.net wrote:

   
  I'm trying to do the same and have read the mentioned sites.  The one
 item I can't seem to get past is a working TFTP server.  What is the
 easiest method to get one running and what packages in Linux or Windows
 work best?

 Thanks for putting up with a Linux newbie.

 Ronny

 -- Original message from Alex Balashov
 abalas...@evaristesys.com: --


 
  Have a look at:
 
 
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080

  094584.shtml#topic2
 
  On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S.
  wrote:
  I understand, but i cannot load the new firmware... is any well
 know
  method?
 
 
  On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
  wrote:
 
  This phone is currently running the SCCP (Skinny) image.
  Before you
  will
  get anywhere you need to load the SIP firmware image onto it.
  The SEP*
  configuration files are for SCCP.
 
  After doing that, the phone will start requesting the correct
 files.
   You
  may need to upgrade through various SIP images cumulatively.
 
  On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S.
 
  wrote:
  Hello I recently get a Cisco 7940G IP Phone and I try to make
 several
  things with it and I en counted many difficulties:
 
  1.) I tried to unlock the phone and to set manually IP Address,
  Netmask, Gateway etc. I don't get any luck.
  2.) I tried to upgrade firmware like they said with tftp
 server... I
  downloaded: P0S3-08-11-00.zip and I uncompressed the files in
 tftpboot
  directory.
  I don't get any luck here either. I look in the
 /var/log/messages and
  I observed that my phone request 4 different files that i
 don't have
  it in my tftp directory.
  Here's my tftp output session with my phone:
 
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to
  192.168.1.3:52178
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
  SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 XMLDefault.cnf.xml
  to 192.168.1.3:52180
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf
  to 192.168.1.3:52181
  Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving
 SEPDefault.cnf to
  192.168.1.3:52182
 
  as you see my phone request 4 files that doesn't comes in
 archive
  P0S3-08-11-00.zip:
  SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml,
 SEP00141CAA4B4C.cnf,
  SEPDefault.cnf...
 
  while my archive contents is the following:
  OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn,
 P0S3-08-11-00.loads,
  P0S3-08-11-00.sb2
 
  3.) I want to make this phone to be SIP compatible. A friend
 of main
  gave me a .cnf file with an example of configuration for SIP.
  How may I rename this cnf file to make work with my phone.
 
  4.) On the other side my phone doesn't have ringtone either.
 Any clue
  how may I put ringtones on it?
 
  I know is a lot of questions for you guys, but I browse on
 cisco.com
  web site and google for hours and I don't get it any clue to
 make work
  this phone in any way.
 
  Thank you for help.
 
  Jonson.
 
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  Alex Balashov
  Evariste Systems
  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (678) 237-1775
 
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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread Ronny Julian

Jeff LaCoursiere wrote:
 If you are running Linux you already have a TFTP server.  You just need to 
 enable it.

   
So there is one on the Asterisk box then?  How would I enable it there?  
Thanks for helping a newbie!


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Re: [asterisk-users] PRI Test Lab

On Fri, Feb 13, 2009 at 07:49:05AM -0600, Kevin P. Fleming wrote:
 Olivier wrote:
 
  What if using a patch cord from port to another ?
  Would you still value to zero each span timing priority or would you set
  one end to 0 and the other to 1 ?
 
 For connecting spans on a single card to each other, all the spans
 should be set to '0'; there is no need to use a recovered clock for any
 spans when they are all on the same card and using the same clock for
 transmission.

But this is not the case if both are on two different cards.
In addition to that, does a T1/E1 link work well when nither side
attempts to set timing?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am

2009-02-13 Thread Anthony Messina

On Friday 13 February 2009 07:54:48 Jeff LaCoursiere wrote:
 Is there a Chicago area users group?  If not is there any interest in
 creating one?

there is: http://groups.google.com/group/asterisk-chicago

though it's fairly inactive.

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E



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Re: [asterisk-users] Cisco IP Phone 7940G.

On Fri, Feb 13, 2009 at 10:23:04AM -0500, John Novack wrote:
 The solar winds windows server is very nice while troubleshooting 
 though, as you have an on screen log that gives real time reports if 
 something is in error.

Hint: 

  tail -f /var/log/daemon.log

(on Debians. probably /var/log/messages instead on RedHats)

If you ever used PXE, you would have noticed that checking for a file
that is not there is a common practice.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Cisco IP Phone 7940G.


On Fri, 13 Feb 2009, Ronny Julian wrote:

 Jeff LaCoursiere wrote:
 If you are running Linux you already have a TFTP server.  You just need to
 enable it.


 So there is one on the Asterisk box then?  How would I enable it there?
 Thanks for helping a newbie!


Depends on your flavor of Linux and what you have installed to manage your 
inbound network connections.  For CentOS/xinetd you will have a directory 
/etc/xinetd.d with a file tftp.  Edit this file and change the line that 
says disable = yes to disable = no.  You then need to send a HUP 
signal to the xinetd process to get it reread these configs.  Get the 
process ID with ps -eaf | grep xinetd, then send it the signal with 
kill -1 pid.

If you don't have xinetd installed you may have regular inetd.  In this 
case edit /etc/inetd.conf, find the line that starts with tftp and make 
sure it isn't commented out.  Send the HUP signal to the inetd process as 
above.

In any case you should see attempts to GET or PUT files in 
/var/log/messages (or /var/log/*tftpd*).

Cheers,

j

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[asterisk-users] Continue processing AGI script after hangup

2009-02-13 Thread cbbs70a


All;
  I wrote a PERL AGI script that prompts a caller to leave a message using 
print RECORD FILE $recordfile wav # 6 BEEP s=3\n; 

When the caller is done, they need to press the # key. The message is then 
delivered.
However, the message is not delivered if the caller simply hangs up when 
finished.
If the user hangs up, the script ends right then. How do I keep on processing 
the
rest of the script after the hangup? Any help at all would be greatly 
appreciated.
Thanks




_
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Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am

2009-02-13 Thread Tim Nelson

I'm just a tad north of you up here in Duluth. However, you just *had* to pick 
Valentines Day for the meetup didn't you? Ugh. I'd love to make the drive and 
come down to discuss Asterisk but unfortunately the *other* love of my life 
wouldn't quite agree with that. :-)

Do you have any sort of site/mailing list/etc setup to facilitate this group? 
I'd be interested in attending such a meetup in the future.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- asterisk help asterisk_h...@iwishi.nu wrote:

 Hello Asterisk Users and those with an Interest in VoIP Tech,
 
 The agenda is open for our next meeting. I think we'll plan on an open
 
 discussion of anyone's choosing. If we're lacking a topic, we'll give
 a 
 demo of installing fail2ban for your asterisk system.
 
 Bring your questions, ideas and projects and we will help you work
 through 
 them.
 
 Jimmy John's is just a block away or we can order pizza. The meeting
 will 
 begin at 11:30am on Saturday.
 
 There is a large parking lot in the rear of the building. Please note,
 the 
 doors at the front of the property are not available for use, you must
 
 enter on the South side of the building (next to the parking area). 
 Building is on the corner of Raymond Ave and University Ave.
 
 The address is 2356 University Ave West. Saint Paul Minnesota, 55114.
 We 
 will meet in suite 401.
 
 We'll have another book to give away as a door pirze, Nagios second
 
 edition.  A special thank you again to No Starch Press and O'Reilly 
 Publishing.
 
 Hope you can make it!
 
 -Eric Osterberg
   Sound Choice Communications LLC
 
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Re: [asterisk-users] Broken Pipe error while using UpdateConfig command

2009-02-13 Thread Örn Arnarson

I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel...

On Fri, Feb 13, 2009 at 8:40 AM, Rilawich Ango maillist...@gmail.comwrote:

 I also experience that problem.  Is it a bug?

 On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com
 wrote:
  Remco Barendse wrote:
  1.4.23.1 is quite badly broken and there are no significant new
  features
 
 
  There are no new features at all, actually. What problems are you having
 with
  1.4.23.1? It doesn't accomplish much to say that it is quite badly
 broken
  without at least telling what is wrong.
 
  We can't fix what's wrong if we don't know what's wrong to begin with. :)
 
  Mark Michelson
 
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Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am

2009-02-13 Thread Shane Young

Quoting Tim Nelson tnel...@rockbochs.com:

 Do you have any sort of site/mailing list/etc setup to facilitate   
 this group? I'd be interested in attending such a meetup in the   
 future.

http://www.tcaug.net/







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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009-02-13 Thread pe...@networkoblivion.com

 1. Is it a technical reason that the ISP has restricted the upload to
 512 Kbps or is it a Marketing reason that they have restricted the
 upload ?

Marketing most likely.  If they have fiber in the building, it would be 
a symmetric link.  I have never seen a fiber connection that isn't the 
same up and down.  They are just trying to sell it like DSL.


 2. Can I boot the cisco switch in run level 1 and modify the rate
 limits on each of the ports ?

2950's don't support rate limiting, they are just semi-dumb layer2 
devices.  They have to be doing this upstream on a router somewhere.

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[asterisk-users] Asterisk and Amazon EC2 cloud service tutorial

2009-02-13 Thread Eric Chamberlain

Voxilla published a two-part series on using Amazon’s cloud services  
to meet your business telephony needs. Part 1 covers Amazon EC2 and  
how it is used in a VoIP setting. Part 2, covers all the steps  
necessary in getting the open-source Asterisk PBX to work on Amazon’s  
cloud.

http://voxilla.com/2009/02/12/amazon-ec2-voip-1096

--
Eric Chamberlain






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Re: [asterisk-users] Continue processing AGI script after hangup

2009-02-13 Thread Tilghman Lesher

On Friday 13 February 2009 09:55:49 cbbs...@hotmail.com wrote:
 All;
   I wrote a PERL AGI script that prompts a caller to leave a message using
 print RECORD FILE $recordfile wav # 6 BEEP s=3\n;

 When the caller is done, they need to press the # key. The message is then
 delivered. However, the message is not delivered if the caller simply hangs
 up when finished. If the user hangs up, the script ends right then. How do
 I keep on processing the rest of the script after the hangup? Any help at
 all would be greatly appreciated. Thanks

$SIG{HUP} = \send_message;

-- 
Tilghman

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[asterisk-users] Asterisk 1.6.0.5 and Aastra phones...

2009-02-13 Thread Carlos Chavez

Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
Making calls is not a problem but when you receive a call it always
drops at 1:45 minutes, always!  

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] PRI Test Lab

Matt Florell wrote:

 Can you tell me where the setting is to force Master timing on Digium
 cards per port? I really didn't think Digium cards had the ability to
 force Master in this way. I've tried to do it with channelbanks before
 and couldn't force it to master, whereas I can get it to work with
 Sangoma.

I'm not sure that the terms you are using match what I'm used to hearing
and telling people, so let me try to describe how our cards work and you
can tell me how that matches up with what you want to do.

Digium multi-port T1/E1 cards use a single clock source for transmitting
on all connected spans. That clock source can be on the onboard
oscillator, or the recovered clock from one of the spans, and the source
selection can change dynamically based on the configuration (in other
words, you can set the first span as 'highest priority', the second span
as 'second priority', etc). If a span is selected to be the clock source
for the entire card, and then that span goes into red alarm, a different
clock source will be chosen, until that span recovers.

So, what this means is that each span port that is configured to be used
is *always* transmitting a bit stream, and due to the nature of T1/E1
signaling, that bit stream includes a clock. A device connected to that
port will *always* be able to recover a clock from that bitstream if it
chooses to do so. For channel banks, other servers, downstream PBXes,
etc. this is a common configuration, and the device will derive its
clock from the bitstream it receives from the Digium card.

In cases where the system admin is using spans that are generated from
'upstream' devices, where the card should slave its transmitted clock to
the recovered clock from that span, then the card can be configured in
this mode. Once a span has been selected to be the clock source for the
card, *all* the spans on that card will use that clock source for their
transmitted bit streams.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] PRI Test Lab

Tzafrir Cohen wrote:

 But this is not the case if both are on two different cards.
 In addition to that, does a T1/E1 link work well when nither side
 attempts to set timing?

You are correct; if the spans are on different cards, then one of them
would ideally be configured to use the clock recovered from the span,
instead of its own onboard clock source. However, even if both cards are
configured to use their own clock sources, if those clocks are close
enough in frequency and don't have appreciable jitter, the system would
still work just fine. It's not an optimal configuration, though.

I don't know that I know what you mean by 'set timing'; if you transmit
a bitstream onto a T1/E1 span, that bitstream includes clocking, by
definition. The source of that clocking is what we are talking about,
not whether it is present or not.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

2009-02-13 Thread John Todd


I've been involved with getting better data for running Asterisk on  
the Amazon EC2 cloud computing system.  Here are some calculations  
I've made on costs based on current published prices on Amazon's  
system.  Feel free to tell me that I'm wrong with these calculations -  
but be specific if you find any problems, as I suspect others may glom  
onto these figures as gospel and I'd hate to have the wrong data in  
there.

   http://www.loligo.com/asterisk/misc/amazon-ec2.xls

The net of my calculations is that a small instance of 20 users in a  
standard office environment would cost about $75 per month, which when  
compared to running a server in-house works out to be (raw cost, not  
including admin time and not discounting out-of-office bandwidth) only  
$38.56 more.  Very interesting.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread David Gibbons

I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.

I'll provide SEPMAC.cnf.xml's if requested off-list.

--Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Friday, February 13, 2009 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco IP Phone 7940G.



Catalin S. wrote:
 hey finally i did it. I upgraded the firmware to the latest sip firmware and 
 now i have the another problem. The requested files are the following:

 ---///---
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf
 to 192.168.1.3:51253
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to
 192.168.1.3:51254
 ---///---

 I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is 
 the mac address of phone, but i don't know what to write in 
 CTLSEP00141CAA4B4C.tlv,
Create an empty file and it will be happy. At least that has been my
experience with my 7960.

Others can probably provide a sample of the remaining files.

So far I have been unable to go beyond version 7 firmware, as it is
unhappy with the XML file when trying to move to version 8.

John Novack

--
Dog is my co-pilot


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Re: [asterisk-users] Asterisk 1.6.0.5 and Aastra phones...

2009-02-13 Thread Anthony Messina

On Friday 13 February 2009 11:39:07 Carlos Chavez wrote:
   Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
 Making calls is not a problem but when you receive a call it always
 drops at 1:45 minutes, always!

I use 1.6.0.5 with 3 Aastra 480i CT phones and have no issues with any of them 
dropping calls from either the base station or the wireless handsets.

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E



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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
 
 I've been involved with getting better data for running Asterisk on  
 the Amazon EC2 cloud computing system.  Here are some calculations  
 I've made on costs based on current published prices on Amazon's  
 system.  Feel free to tell me that I'm wrong with these calculations -  
 but be specific if you find any problems, as I suspect others may glom  
 onto these figures as gospel and I'd hate to have the wrong data in  
 there.
 
http://www.loligo.com/asterisk/misc/amazon-ec2.xls
 
 The net of my calculations is that a small instance of 20 users in a  
 standard office environment would cost about $75 per month, which when  
 compared to running a server in-house works out to be (raw cost, not  
 including admin time and not discounting out-of-office bandwidth) only  
 $38.56 more.  Very interesting.

For 20$ or slightly more you can rent a Xen or OpenVZ virtual host which
will probably do as well.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread Anthony Francis



David Gibbons wrote:
 I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.

 I'll provide SEPMAC.cnf.xml's if requested off-list.

 --Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
 Sent: Friday, February 13, 2009 9:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco IP Phone 7940G.



 Catalin S. wrote:
   
 hey finally i did it. I upgraded the firmware to the latest sip firmware and 
 now i have the another problem. The requested files are the following:

 ---///---
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
 SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf
 to 192.168.1.3:51253
 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to
 192.168.1.3:51254
 ---///---

 I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is 
 the mac address of phone, but i don't know what to write in 
 CTLSEP00141CAA4B4C.tlv,
 
 Create an empty file and it will be happy. At least that has been my
 experience with my 7960.

 Others can probably provide a sample of the remaining files.

 So far I have been unable to go beyond version 7 firmware, as it is
 unhappy with the XML file when trying to move to version 8.

 John Novack

 --
 Dog is my co-pilot

   
On a similar subject, I have been able to get a 7961 to switch to a SIP 
firmware, has anyone had any luck with this?

-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP



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[asterisk-users] OpenSky: Digium Skype gateway?

2009-02-13 Thread Philipp von Klitzing

Hi there,

is gizmo the first user of the Digium Skype solution, or do they use a 
different approach/product - any clue?

http://www.gizmo5.com/pc/opensky/

Philipp


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Re: [asterisk-users] PRI Test Lab

On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
 Matt Florell wrote:

   Can you tell me where the setting is to force Master timing on Digium
   cards per port? I really didn't think Digium cards had the ability to
   force Master in this way. I've tried to do it with channelbanks before
   and couldn't force it to master, whereas I can get it to work with
   Sangoma.


 I'm not sure that the terms you are using match what I'm used to hearing
  and telling people, so let me try to describe how our cards work and you
  can tell me how that matches up with what you want to do.

  Digium multi-port T1/E1 cards use a single clock source for transmitting
  on all connected spans. That clock source can be on the onboard
  oscillator, or the recovered clock from one of the spans, and the source
  selection can change dynamically based on the configuration (in other
  words, you can set the first span as 'highest priority', the second span
  as 'second priority', etc). If a span is selected to be the clock source
  for the entire card, and then that span goes into red alarm, a different
  clock source will be chosen, until that span recovers.

  So, what this means is that each span port that is configured to be used
  is *always* transmitting a bit stream, and due to the nature of T1/E1
  signaling, that bit stream includes a clock. A device connected to that
  port will *always* be able to recover a clock from that bitstream if it
  chooses to do so. For channel banks, other servers, downstream PBXes,
  etc. this is a common configuration, and the device will derive its
  clock from the bitstream it receives from the Digium card.

  In cases where the system admin is using spans that are generated from
  'upstream' devices, where the card should slave its transmitted clock to
  the recovered clock from that span, then the card can be configured in
  this mode. Once a span has been selected to be the clock source for the
  card, *all* the spans on that card will use that clock source for their
  transmitted bit streams.

I guess I'm not explaining myself very well, so I'll describe exactly
what happened and how my problem was solved. We had a digium quad port
T1 card with 3 carrier T1s plugged into it and one channelbank. After
a few months of everything running just fine on the system the
channelbank would go red alarm after a few hours of the server being
on, if we reset the server, channelbank or even just unplugged the
crossover T1 and plugged it back into the channelbank it would work
again for a few more hours. The carrier told us it was a timing issue,
so I began to mess with the timing settings and after a week of making
changes and waiting to see if they would work, none of the changes to
any of the timing settings in zapata.conf would do anything. At this
point I swapped out the Digium card with a Sangoma card(because the
quad T1 cards were cheaper than a new channelbank) and the same thing
happened. I emailed Sangoma support and they suggested I try the
forced Master clock setting for the channelbank port to see if that
would help, and after figuring out exactly how to set it up I put it
live and no more red alarms on the channelbank.

My understanding is that this setting lets you ignore the timing
signal coming from the other end of one of the ports, and the card
will take another timing source from a port that you specify and
force it to be used as a timer on that first port.

If my explanation makes no sense I apologize, but this is how it
happened and the problem was solved.

MATT---

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Re: [asterisk-users] OpenSky: Digium Skype gateway?

2009-02-13 Thread randulo

On Fri, Feb 13, 2009 at 8:19 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de  is gizmo the first user of
the Digium Skype solution, or do they use a
 different approach/product - any clue?

No clue how it's done, but it cuts off at 5 minutes and the quality
isn't fantastic

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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

2009-02-13 Thread John Todd


On Feb 13, 2009, at 11:05 AM, Tzafrir Cohen wrote:

 On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:

 I've been involved with getting better data for running Asterisk on
 the Amazon EC2 cloud computing system.  Here are some calculations
 I've made on costs based on current published prices on Amazon's
 system.  Feel free to tell me that I'm wrong with these  
 calculations -
 but be specific if you find any problems, as I suspect others may  
 glom
 onto these figures as gospel and I'd hate to have the wrong data in
 there.

   http://www.loligo.com/asterisk/misc/amazon-ec2.xls

 The net of my calculations is that a small instance of 20 users in a
 standard office environment would cost about $75 per month, which  
 when
 compared to running a server in-house works out to be (raw cost, not
 including admin time and not discounting out-of-office bandwidth)  
 only
 $38.56 more.  Very interesting.

 For 20$ or slightly more you can rent a Xen or OpenVZ virtual host  
 which
 will probably do as well.


For the average Xen VM host, what's the price for bandwidth, assuming  
G.711 channels at the rates I quoted?  I don't have a list of  
bandwidth costs here for other hosting providers.   Of course, Amazon  
is simply the most well-known cloud system at the moment but you can  
fill in the slots in the spreadsheet with whatever data might be  
applicable.  Amazon also has a decent IP network peering structure  
from what I've seen, so paths into their network are typically short  
and direct.  How packets move around inside their network is still up  
to experimentation - waiting on someone to test jitter, packet loss  
(coughRONALD-AND-NIRcough) and other criteria.

The wildcard in the calculation (which is not examined) is having  
enough bandwidth out of your office to handle X simultaneous calls,  
and how much that costs.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] OpenSky: Digium Skype gateway?

2009-02-13 Thread John Todd


On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:

 Hi there,

 is gizmo the first user of the Digium Skype solution, or do they use a
 different approach/product - any clue?

 http://www.gizmo5.com/pc/opensky/

 Philipp


I know nothing about their solution that I can say with assurance  
other than it's not the Digium/Skype gateway software.

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Cisco IP Phone 7940G.

2009-02-13 Thread David Gibbons

snip
On a similar subject, I have been able to get a 7961 to switch to a SIP
firmware, has anyone had any luck with this?
/snip

Yes, I have several 7961s and 7971s running SIP, same firmware generation as 
the 41s

--Dave

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Re: [asterisk-users] PRI Test Lab

Matt Florell wrote:

 My understanding is that this setting lets you ignore the timing
 signal coming from the other end of one of the ports, and the card
 will take another timing source from a port that you specify and
 force it to be used as a timer on that first port.

That is very close to what our cards do, except that it's not
controllable on a port-by-port basis.

In that situation, which port did you pick as the clock source for the
channelbank port, and what was the clock source for it prior to that?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

On Fri, Feb 13, 2009 at 11:35:53AM -0800, John Todd wrote:
 
 On Feb 13, 2009, at 11:05 AM, Tzafrir Cohen wrote:
 
  On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
 
  I've been involved with getting better data for running Asterisk on
  the Amazon EC2 cloud computing system.  Here are some calculations
  I've made on costs based on current published prices on Amazon's
  system.  Feel free to tell me that I'm wrong with these  
  calculations -
  but be specific if you find any problems, as I suspect others may  
  glom
  onto these figures as gospel and I'd hate to have the wrong data in
  there.
 
http://www.loligo.com/asterisk/misc/amazon-ec2.xls
 
  The net of my calculations is that a small instance of 20 users in a
  standard office environment would cost about $75 per month, which  
  when
  compared to running a server in-house works out to be (raw cost, not
  including admin time and not discounting out-of-office bandwidth)  
  only
  $38.56 more.  Very interesting.
 
  For 20$ or slightly more you can rent a Xen or OpenVZ virtual host  
  which
  will probably do as well.
 
 
 For the average Xen VM host, what's the price for bandwidth, assuming  
 G.711 channels at the rates I quoted?  

I recently got myself a really cheap Xen hosting (10$ a month). It
includes a bandwith usage of up to 2mbps . Note that I don't intend to
use it for VoIP hosting.

There are plenty of providers. You need to shop around a bit.

 I don't have a list of  
 bandwidth costs here for other hosting providers.   Of course, Amazon  
 is simply the most well-known cloud system at the moment but you can  
 fill in the slots in the spreadsheet with whatever data might be  
 applicable.  Amazon also has a decent IP network peering structure  
 from what I've seen, so paths into their network are typically short  
 and direct.  

The location of the hosting facility is something you can easily find
with every such provider.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] linksys PAP2t and asterisk

2009-02-13 Thread wassim Darwish


Hi all:
when i make a call from linksys pap2t to an asterisk server a fake ring is 
heard some times ,but when sending calls between 2 asterisk servers through sip 
no fake ring is heard but real one. 
any suggestions please.
 

_
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Re: [asterisk-users] PRI Test Lab

On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
 Matt Florell wrote:

   My understanding is that this setting lets you ignore the timing
   signal coming from the other end of one of the ports, and the card
   will take another timing source from a port that you specify and
   force it to be used as a timer on that first port.


 That is very close to what our cards do, except that it's not
  controllable on a port-by-port basis.

  In that situation, which port did you pick as the clock source for the
  channelbank port, and what was the clock source for it prior to that?

When I got it working I had set the clock source to the 4th port on
the card for the channelbank which was on the first port. We never
tried any other ports because I was afraid to touch it after I got it
working.

Before it was working, we had tried every one of the dozens of
combinations of timing settings in zapata.conf for the 4 spans, and
none of them worked.

As a note, I should mention that this was over 2 years ago.

Thanks,

MATT---

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[asterisk-users] GUI interface to manage business edition

2009-02-13 Thread Miguel Martinez

I am new to the Asterisk world, but have decided to use the business
edition, but am looking for a cost effective gui interface to manage the
software.  I have heard of FreePBX, ScopServ and Trixbox from Fonality.  Can
anyone give me a heads-up on any of these products and possibly a
comparison?  

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Re: [asterisk-users] PRI Test Lab

Matt Florell wrote:

 When I got it working I had set the clock source to the 4th port on
 the card for the channelbank which was on the first port. We never
 tried any other ports because I was afraid to touch it after I got it
 working.

Understandable; as far as I know, the Digium card should have been able
to do what you needed, and it could very well have been driver bugs or
something else like that which kept it from working back then :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Cisco IP Phone 7940G.

i finally did it... It works excellent. Thank you guys for help.


On Fri, Feb 13, 2009 at 9:44 PM, David Gibbons d...@videon-central.com wrote:
 snip
 On a similar subject, I have been able to get a 7961 to switch to a SIP
 firmware, has anyone had any luck with this?
 /snip

 Yes, I have several 7961s and 7971s running SIP, same firmware generation as 
 the 41s

 --Dave

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[asterisk-users] Asterisk 1.6.0.6 Release Candidate 1 Now Available

2009-02-13 Thread Asterisk Development Team

The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 1.6.0.6, tagged as version 1.6.0.6-rc1. Release candidate
1.6.0.6-rc1 is available for immediate download at http://downloads.digium.com/.

This release of Asterisk fixes several issues related to overall stability and
usability, especially with applications and their options. In addition,
an issue with IAX2 transfers not taking place has been fixed. Issues found in
this release candidate can be reported at http://bugs.digium.com/.

For a full list of changes in this release candidate, please see the
ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.6.0.6-rc1/ChangeLog?view=co

Thank you for your continued support of Asterisk!


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Re: [asterisk-users] Asterisk 1.6.0.5 and Aastra phones...

2009-02-13 Thread Carlos Chavez

Could you please send me a copy of your sip.conf file so I can compare
the settings?  Aastra 5xi phones in my office always drop the call after
105 seconds and older type phones like the 9133i and 480i crash after
about a minute.  It only happens with Aastra phones and only when
receiving calls.

On Fri, 2009-02-13 at 12:56 -0600, Anthony Messina wrote:
 On Friday 13 February 2009 11:39:07 Carlos Chavez wrote:
Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
  Making calls is not a problem but when you receive a call it always
  drops at 1:45 minutes, always!
 
 I use 1.6.0.5 with 3 Aastra 480i CT phones and have no issues with any of 
 them 
 dropping calls from either the base station or the wireless handsets.
 
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-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] GUI interface to manage business edition

2009-02-13 Thread Bob Pierce


On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote:
 I am new to the Asterisk world, but have decided to use the business
 edition, but am looking for a cost effective gui interface to manage
 the software. 

Does the Asterisk-GUI work with Asterisk Business Edition?


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Re: [asterisk-users] GUI interface to manage business edition