date:20061008



On 09/10/2006, at 12:12 PM, Dean Collins wrote:


can anyone give me an idea on how this reboot script works?


I actually just use the SIP notify command on the Asterisk console to  
remotely reboot my Polycom phones. It requires a pre-configured  
sip_notify.conf file and the Polycom option to reboot on config  
check. You can then call it from a script using:


# asterisk -rx "sip notify polycom-reboot 400"

(Where 400 is the SIP ID of the phone).

I'm interstate at the moment, but if you send me an email, I can  
lookup the settings when I'm back on Wednesday.


Ta,
Avi

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RE: [asterisk-users] ftp server


On Mon, October 9, 2006 11:46 am, Dean Collins said:
> Are you able to track real time from a windows machine the transactions
> occurring on your asterisk server if you have vsftpf installed?

Yes... In an SSH session, "tail -f /var/log/vsftpd.log" will show you
everything you need.

Also, I have all my .cfg files in one directory, and then
each phone gets its own directory to store its own phone.cfg and log
files. Works like a charm.

cYa,
Avi


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Re: [asterisk-users] DID is not working (call is not routing)

2006-10-08 Thread William Piper

Your server seems to be doing exactly what you are telling it to do:
 
 -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') 
Read the extensions.conf directions on the wiki site:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
 
bp 
On 10/8/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.
When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: 
*CLI>    -- IAX2/teliax-2 answered SIP/350-09e3b540    -- Executing GotoIf("SIP/216.89.79.2 
-09e1d020", "0?from-trunk||1") in new stack    -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack     -- Channel will hangup at 2006-10-06 11:27:55 UTC.
    -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack    -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack     -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')    -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack   == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020'
    -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack    -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack     -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack
    -- Goto (from-sip-external,s,1)    -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack     -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2006-10-06 11:28:04 UTC.    -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack   == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'
When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully.  Please tell me the solution. Looking forward to your response. Thank you. 
Regards,Chandra.


Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. 
Great rates starting at 1¢/min. 
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Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-08 Thread Kristian Kielhofner

Jeremy McNamara wrote:

Tzafrir Cohen wrote:
> H, I'm not sure that this is exactly the data you're after.

You're looking for the ammounts of writes for the disk block that gets
the most writes.

E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the system would remain
mostly unchanged.

Again, if the embedded system is setup properly, there is NO writing to
the flash during normal operations, thus the device won't be killed by
its alleged 2 million write limitation.

Kris and I had a quick discussion on this topic, off-list, and his
original flash-based device is still in constant operation after 2 years
and I have flash modules that I purposely tried to kill with writes. It
took significant effort to start causing error situations, which were
very easily detected before the system would become unusable.

Erick, you should focus on having a quick action restoration plan and
extra DOMs always readily available. Then when a failure situation is
detected, you can react very quickly.

Jeremy McNamara

Jeremy, Erick -

I have always pointed to this SanDisk whitepaper:

http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf

While it specifically discusses their industrial line of CF cards, it
is pretty obvious that flash can, and often does, last much longer than
other components in a system when properly implemented. You will notice
that the SHORTEST expected life of a CF card in their test scenarios was
over 70 years! How long is your power supply going to last? Even if
the consumer level cards had 1/10 the life expectancy, that is still
seven years. I expect to get at least that from my original AstLinux
system. It's been two so far, I'll let you know how it is doing in
another five years :).

JFFS (and similar FSs) are not appropriate for CF cards or DOMs. They
are meant to be used directly on flash memory and do their own wear
leveling and in some cases, compression. All kinds of commercial
devices use JFFS2. If you are using a CF or DOM with Linux, ext2 is the
best FS to use. CF cards and DOMs use their own wear leveling, so none
is required in the operating system or file system. CF cards and DOMs
hide wear leveling from you and expose themselves as an ordinary IDE device.

I echo Jeremy's conclusions. With a properly designed operating
system, decent flash memory, and a reasonable usage pattern, I can tell
you (with a great amount of certainty) that in most situations, CF cards
will outlast just about any hard drive (even SCSI) when used 24/7.
These days, it really is pretty tough to trash flash.

However, if you are running a MySQL cluster or something with several,
multi-gigabyte databases, no type of flash memory will last very long! :)

To get back to answering your question, I HIGHLY recommend that you
avoid MySQL and realtime on your box with a DOM. Nothing against either
(MySQL or Realtime), but they will probably make your device more
complicated than it needs to be while substantially shortening the life
of your DOM. If you absolutely have to use MySQL, you might have better
luck using a MySQL storage engine that uses fewer writes than InnoDB,
but I am no expert on that.

--
Kristian Kielhofner
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[asterisk-users] polycom reboot script









http://www.freedomphones.net/polycom/files/PolyReboot.pl-script

 

can anyone give me an idea on how this reboot script works?

 

How do I load it up and cause it to occur etc? sorry for
being a newbie but searching on perl script information on google isn’t
working.

 

 

 

Cheers,

 

Dean

 

 






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Re: [asterisk-users] Re: Real-time and priority "n"

2006-10-08 Thread Ronald Wiplinger

Brian Capouch wrote:

Tony Mountifield wrote:

In article <[EMAIL PROTECTED]>,
Ronald Wiplinger <[EMAIL PROTECTED]> wrote:

Is it exclusive? Either Realtime or priority "n" ???

If so, what is the better way?

I believe 'n' is just a shorthand way of writing "previous line + 1",
and gets converted into an actual number as the dialplan is compiled.
After compilation, the information about whether a line had been given
as 'n' or as a specific number has been lost, as far as I know.

Rows can be added to a database table at any time. Imagine a series
of priorities added to a table using nothing more than "n" as a
priority number beyond the first one.

Now imagine wanting to add a new priority in between any two arbitrary
entries in the table. How would you even specify which two lines
should surround it, when they have no identifying "serial number"
associated with them?

Unless you were to add a new field, e.g. "priority location
identifier," or somesuch. Which does nothing more than move back to
the present situation.

The extensions.conf parser adds a "real" priority to each line, but in
Realtime that responsibility falls on the DB maintainer.

Short: EXCLUSIVE

thanks!

bye

Ronald

--
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http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com
Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

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If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please.
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.

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[asterisk-users] SIP vs. SIP-B

2006-10-08 Thread Dovid B




Tried looking around (ok maybe not well cause I am 
a lil tired) but cant seem to find it. Can some one send me a link with the 
diffrences between SIP and SIP-B ? Thanks.
 
Dovid
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RE: [asterisk-users] ftp server









Are you able to track real time from a
windows machine the transactions occurring on your asterisk server if you have
vsftpf installed?

 



 

Cheers,

 

Dean

 













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dovid B
Sent: Sunday, 8 October 2006 9:04
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] ftp
server



 



vsftpd.





i use seperate user id's for the phones (makes it easier
than to have all the configs in one dir). Works wonders. and for all you
windows lovers. Time to make the switch







- Original Message - 





From: Dean Collins 





To: asterisk-users@lists.digium.com 





Sent: Sunday, October
08, 2006 7:30 AM





Subject: [asterisk-users]
ftp server





 



Whats the best ftp server to upload Polycom phone
cfg’s from? I’m finding it a bit hit and miss using BTF server.

 

 

 

Cheers,

 

Dean

 

 







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[asterisk-users] Blacklist to check http://whocalled.us

2006-10-08 Thread Noah Swint

There's a product below on the market that checks whocalled.us to
determine if a telmarketer should get the Zapteller.  Do you know if
that's something that could possibly be included into the blacklist or
in a macro.

http://venotec.com/product/tms/


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Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-08 Thread Jeremy McNamara


Tzafrir Cohen wrote:
 > H, I'm not sure that this is exactly the data you're after.


You're looking for the ammounts of writes for the disk block that gets
the most writes.

E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the system would remain
mostly unchanged.



Again, if the embedded system is setup properly, there is NO writing to 
the flash during normal operations, thus the device won't be killed by 
its alleged 2 million write limitation.


Kris and I had a quick discussion on this topic, off-list, and his 
original flash-based device is still in constant operation after 2 years 
and I have flash modules that I purposely tried to kill with writes. It 
took significant effort to start causing error situations, which were 
very easily detected before the system would become unusable.


Erick, you should focus on having a quick action restoration plan and 
extra DOMs always readily available.  Then when a failure situation is 
detected, you can react very quickly.





Jeremy McNamara


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Re: [asterisk-users] 2x* and realtime

2006-10-08 Thread Dovid B

Doug asked a while back if there was a way to share astdb between two 
machines. If there was a way then the registry would be the same. While on 
the topic I may be completly off however if two machines were to share the 
same real time db as well as the same astdb (assuming we can share it 
between two machines) wouldnt that be a safe way to run asterisk with 
failover (asuming VOIP only).


Dovid
- Original Message - 
From: "Rushowr" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Sunday, October 08, 2006 10:12 PM
Subject: Re: [asterisk-users] 2x* and realtime



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Re: [asterisk-users] ftp server

2006-10-08 Thread Dovid B




vsftpd.
i use seperate user id's for the phones (makes it 
easier than to have all the configs in one dir). Works wonders. and for all you 
windows lovers. Time to make the switch

  - Original Message - 
  From: 
  Dean 
  Collins 
  To: asterisk-users@lists.digium.com 
  
  Sent: Sunday, October 08, 2006 7:30 
  AM
  Subject: [asterisk-users] ftp 
server
  
  
  Whats the best ftp server to 
  upload Polycom phone cfgs from? Im finding it a bit hit and miss using BTF 
  server.
   
   
   
  Cheers,
   
  Dean
   
   
  
  

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Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

On Fri, Oct 06, 2006 at 05:10:34PM -0500, Erick Perez wrote:
> I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!)
> 
> However the main question was not aswered (or i didn't get it, did I ?)
> 
> If I use a Disk on Module that has 2million hours MTBF and a Read/Write
> lifecycle of 2million times, then, How many days/weeks/months/years will
> take to do 2million read/write cycles?
> which leads to my second question.
> How do I measure/count the read and writes a normal linux system running
> asterisk does during a day, so I can extrapolate that in terms of time? Is
> there an utility?
> 
> Example: if I setup system XYZ with asterisk, then load this magical
> utility/procedure that counts how many writes the filesystem has done to /
> or to /,/tmp,/var and after 24 hours the utility/procedure says: 10thousand
> writes, then, I will do
> 10thousand writes a day multiplied by 200 days = 2 millions
> Obviously this means I will not use a RAM disk and I want to write to the
> module everytime
> 
> Then i will assume that the Disk on a Module will die after 200 days. Or am
> I completely and horribly misunderstanding the "2million
> Read/Write LifeCyle" advertised by Disk-on-Module companies?
> 
> Example:
> http://www.pqi.com.tw/product2.asp?oid=140&cate1=143&PROID=34
> ‧MTBF:2,000,000 Hours
> ‧R/W Cycle:2,000,000 Times

H, I'm not sure that this is exactly the data you're after.

You're looking for the ammounts of writes for the disk block that gets
the most writes.

E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the system would remain
mostly unchanged.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] ftp server

2006-10-08 Thread Robert Goodyear

Whoops. I totally generalized under the realm of "Non Windows" didn't  
I? Doh!


On Oct 8, 2006, at 2:27 PM, Tzafrir Cohen wrote:


On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote:


On Linux, I've fallen in love with FreeNAS, and not just because it
utilizes the m0n0wall GUI. You can boot it off a USB stick, too,
which I find super-cool.
http://www.freenas.org


/me imagines some BSD fanatics readying their tridents and aiming at
Robert :-)



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Re: [asterisk-users] Optus PRI via DSL



On 08/10/2006, at 9:34 PM, Paul Hales wrote:

I have seen an Optus SHDSL box set up incorrectly before - and the  
tech

re-visited and set it up correctly within hours of being informed.


Same with my Optus SHDSL box: The first tech misconfigured, so I kept  
getting PRI restarts on my Sangoma card. They came back and  
reconfigured and now it works like a charm.


cYa,
Avi

--
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< Sydney / Melbourne / Canberra / Hobart / London />
  2/340 Gore StreetT: +61 (0) 3 9235 5400
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Re: [asterisk-users] ftp server



On 09/10/2006, at 5:07 AM, Noah Miller wrote:


username and password is "PlcmSpIp".  vsftpd cannot handle capitalized
usernames, so if you want to use vsftpd, you have to manually
re-configure the username on each phone.


I use vsftpd and I'm using the default PlcmSpIp username just  
fine. :) Essentially, I configured PlcmSpIp as a Linux user and I'm  
serving it out by using personalised FTP home directories in vsftp  
and then chrooting per user. Works like a charm and no phone  
configuration is required.


cYa,
Avi

--
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< Sydney / Melbourne / Canberra / Hobart / London />
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Re: RE : [asterisk-users] New Asterisk StumbleUpon Group

2006-10-08 Thread Matt Riddell (IT)

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

[EMAIL PROTECTED] wrote:
> Hello Matt,
> 
> I have not seen how to add a site.
> Could you help me (us) ?
> Tks

When you are in the site list:

Click the link:

http://asterisk.group.stumbleupon.com/sites/

It's titled, View/Add Sites

- --
Cheers,

Matt Riddell
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFKXpYS6d5vy0jeVcRAsh+AJkBLk+Zfi8bIw3ViyBavjCLwTFozgCeKcPu
7eLlRoM9wAxydb2yX6njjuI=
=lii4
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Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-08 Thread Jay R. Ashworth

On Fri, Oct 06, 2006 at 08:35:26AM -0800, Mojo with Horan & Company, LLC wrote:
> I'm pretty sure that when you AREN'T sending the DTMF inband, asterisk 
> detects it, and if the keys pressed don't lead to any recording/transfer 
> features, then it re-creates DTMF on the bridged channel.  I mean to 
> say, my called party can't hear me start recording or transfer them, but 
> I don't have any trouble with outside IVRs.

So, you're suggesting that the FXO channel driver generates outbound
DTMF under the command of (eventually) the phone set?  That would be
nice.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-08 Thread Erick Perez

I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!)
 
However the main question was not aswered (or i didn't get it, did I ?)
 
If I use a Disk on Module that has 2million hours MTBF and a Read/Write lifecycle of 2million times, then, How many days/weeks/months/years will take to do 2million read/write cycles?
which leads to my second question.
How do I measure/count the read and writes a normal linux system running asterisk does during a day, so I can extrapolate that in terms of time? Is there an utility?
 
Example: if I setup system XYZ with asterisk, then load this magical utility/procedure that counts how many writes the filesystem has done to / or to /,/tmp,/var and after 24 hours the utility/procedure says: 10thousand writes, then, I will do

10thousand writes a day multiplied by 200 days = 2 millions
Obviously this means I will not use a RAM disk and I want to write to the module everytime
 
Then i will assume that the Disk on a Module will die after 200 days. Or am I completely and horribly misunderstanding the "2million Read/Write LifeCyle" advertised by Disk-on-Module companies?
 
Example:
http://www.pqi.com.tw/product2.asp?oid=140&cate1=143&PROID=34
‧MTBF:2,000,000 Hours‧R/W Cycle:2,000,000 Times 
I want to understand if that's what they mean.
 
I fully understand that such media will have a longer life cycle if i only read from it and keep writes to a mimimum, for example: writing dialpan changes.
 
The whole idea comes from doing a mini itx with no moving parts offering voicemail stored in a disk-on-module and astlinux in a CF and a RAM Disk large enough to do processing on RAM before saving to CF or to disk-on-module when needed.

 
Thanks again for you comments,
 
   
On 10/6/06, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
Kristian Kielhofner wrote:>> Erick,>> Or  Just use AstLinux which kind of does what Jeremy described :)
>> http://www.astlinux.org>>> P.S. - I am the creator of AstLinux>> --> Kristian Kielhofner   Sorry to reply to my own post, but there seems to have been some
confusion in what I said here.  To completely clear it up, Astlinux onlywrites to flash in these circumstances:1)  You update the configs.2)  You update AstLinux.3)  You are using voicemail and people leave voicemail. (most flash
seems to last "long enough" given typical voicemail usage patterns)4)  If you have the PERSISTLOG option enabled, I will save syslogs toflash (not RAM - the default).  Users are warned about this, and it is
not the default.5)  astdb is stored in flash, so depending on your needs, SIPregistrations and/or dundi keys may get written here periodically.  Imight make an option similar to PERSISTLOG to disable this.
   Also, you have the option of using a hard drive or alternate flashdevice for ALL writes.  Boot from flash, run from HD.  Do whatever worksbest for you and your application.--Kristian Kielhofner
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Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 
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[asterisk-users] Odd echo issue with speaker phone

2006-10-08 Thread BerkHolz, Steven




I assume that this 
is from the echo canceller, but I am not sure.
 
A call is started 
via SIP speakerphone.
When the handset is 
picked up, there is a slight echo of your own voice after you speak.(duh, is 
there any other kind of echo)
 
If the call is made 
without the speaker phone, there is no echo.
 
I am stumped by this 
one.
 

 
Thank You,
 
Board member 
ofwww.glimasoutheast.org
 
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Re: [asterisk-users] Re: Asterisk Server : IDE HDD frequent crash

2006-10-08 Thread Jay R. Ashworth

On Fri, Oct 06, 2006 at 12:31:24PM -0700, Martin Joseph wrote:
> >On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
> >>Heat = #1 cause of disk failure. If they are roasting to the touch they 
> >>will fail in 2-3 months.
> >
> >One word: "smartd".
> >
> >I didn't know it existed, and I'm amazed I didn't.  Everyone on this
> >list should be running smartd, and know what it's saying.
> 
> SMART is useful, but not the be all and end all of disk drive care.

Necessary, but not sufficient, certainly.

> Proper ventilation as already mentioned, is much more important then 
> SMART status in my opinion...

Oh, of course.  I didn't bother to reinforce that, as I figured
everybody knew that.  :-)

> I have seen many drives that fail, while still reporting that 
> everything is hunk dory as far as "SMART" is concerned.

Really?  Bummer.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
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Re: [asterisk-users] ftp server

On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote:

> On Linux, I've fallen in love with FreeNAS, and not just because it  
> utilizes the m0n0wall GUI. You can boot it off a USB stick, too,  
> which I find super-cool.
> http://www.freenas.org

/me imagines some BSD fanatics readying their tridents and aiming at 
Robert :-)

As quick lok into http://www.freenas.org/ would tell, it runs on FreeBSD
(just like m0n0wall).

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Tellabs and a PRI

2006-10-08 Thread Doug Lytle


Eric "ManxPower" Wieling wrote:
I had a similar problem.  Turned out I wired it backwards.  The 
Tellabs only does EC in one direction.




The Tellabs that I have do Sender side EC as well.  But, I plan on being 
on site next weekend and will try switching the wiring around.


Thanks for the input,

Doug


-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Tellabs and a PRI

2006-10-08 Thread Eric \"ManxPower\" Wieling

I had a similar problem.  Turned out I wired it backwards.  The Tellabs 
only does EC in one direction.


Doug Lytle wrote:

Another question,

Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our 
analog lines to a PRI, I thought it would be simple stuff moving the EC 
to the PRI. Changed signaling, made sure that channel 24 wasn't being 
ECd and everything came up. But, I was getting complaints of random echo 
on the PRI. Local echo. Also, we weren’t able to do any kind of modem 
dial-outs (Adit 600 supplying the dialtone). Funny thing is that 
inbound/outbound faxing was working fine.


Removed the Tellabs and turned on the software EC and everything started 
working fine. I'd really like to move back to the Tellabs, so if anybody 
has suggestions, please speak up.


Forgot to mention in the last post, I'm using a Sangoma A102 with the 
current Beta 7 drivers.

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Re: [asterisk-users] disabling hardware echo can on tdm2400p

2006-10-08 Thread Eric \"ManxPower\" Wieling


Sean Kennedy wrote:

Hey list,

Short version:
I have a need to disable the hardware can on the tdm24xxp I have.  I 
figure it's something in zconfig.h in the zaptel directory, but I'll be 
damned if I can figure it out.


Long version:
I have a tdm2403e card which is experiencing an odd problem;  When 
several lines are in use, there is a "bleeding" of lines.  My users call 
it the 'ghost'.  Regardless, they can hear other people's conversation 
on different lines.  I've been told this has to do with the hardware 
echo can I have on there, and that I should disable it if I continue 
having problems.  So that's where I stand.


Answers and opinions welcome.


Crosstalk is not an Echo Cancel issue.  It is usually caused by a wiring 
issue.  Search the archives or look at the Zaptel README for the module 
option to disable the echo canceler.

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Re: [asterisk-users] 2x* and realtime

2006-10-08 Thread Rushowr


> Is there a way to check if a peer is registered with the other box and
> forward the call there if a call comes in?

Yes, you can (if nothing else, I'm fuzzy this morning) try forwarding
the call and it will fail if the device is not registered because
Asterisk will report it "not found" with a SIP 404



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Re: [asterisk-users] ftp server

2006-10-08 Thread Noah Miller


Hi Dean -


I'm using vsftpd quite successfully on several Asterisk boxes with
Polycom IP501 phones.


Just an addition: one requirement I had in deploying Polycom phones -
I wanted the user (and me) to be able to plug in a new phone and go
with no configuration needed on the Polycom end.  The default FTP
username and password is "PlcmSpIp".  vsftpd cannot handle capitalized
usernames, so if you want to use vsftpd, you have to manually
re-configure the username on each phone.

I chose to use ProFTPd instead, which, with some minor config changes,
can handle the default Polycom username with capital letters.



Btw I just noticed there is no bootrom.ver file in this zip folder.


I think they did away with this file as of the 2.6.2 version.  I just
reviewed the zip files I got from my retailer and 2.6.1 and below have
this file, but the newer ones don't.


- Noah
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Re: [asterisk-users] ftp server

2006-10-08 Thread Robert Goodyear

Couple suggestions:On Win32, try Serv-U FTP. It's very reliable and supports a variety of protocols like SFTP and the like. Commercial software.http://www.rhinosoft.comOn Linux, I've fallen in love with FreeNAS, and not just because it utilizes the m0n0wall GUI. You can boot it off a USB stick, too, which I find super-cool.http://www.freenas.orgGood luck,-Rob.On Oct 8, 2006, at 11:29 AM, Gary Eck wrote: Yes, we don't have the proper skillset, either. Even though we are a reseller.   We have standardized on Linux appliances called Snapgear .  Of course, they are not integrated into Active directory, so there are lots of things that are harder to do.   However, since they are not Microsoft or Cisco based, they are not the targets of many of the severe attacks on the Internet. We also have problems with installing all the security updates on the scores of Small Business Customers we have - they don't want to pay us for installing these updates - much less, paying us when a security update has side effects on their internal network.   So, avoiding the whole Microsoft issue is a decent compromise in our situation.   -Original Message-From:   [EMAIL PROTECTED]   [mailto:[EMAIL PROTECTED]] On Behalf Of Dean   CollinsSent: Sunday, October 08, 2006 1:16 PMTo:   Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:   [asterisk-users] ftp server  Unfortunately I   disagree with you about ISA, whilst it may cause problems through my lack of   skillset from time to time the functionality it introduces and protects my   network cant be beat at any price, being built into sbs 2003 is just a   bonus.     Cheers, Dean   From:   [EMAIL PROTECTED]   [mailto:[EMAIL PROTECTED]] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:37   PMTo: Asterisk Users Mailing   List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp   server   We also have SBS2003   - we initially had ISA running on it, but decided it was not worth the grief   at our office.  In the back of my   mind, I was wondering if ISA was causing the   problem.     I had used   Bulletproof in the past, since it has bandwidth throttling - and it just seems   to work fine.  Unfortunately, it   does not run as a service - but I found another product that allows it run as   a service.  -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp serverYep, using sbs 2003 here.  Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp server We had some problems with the FTP server on our Windows 2003 server - we worked with it quite a bit, but just could not get the Pollycom's to access it. It did work with backup up our Switchvox PBX. We use Bulletproof FTP server in the Windows world. -Original   Message-From:   [EMAIL PROTECTED]   [mailto:[EMAIL PROTECTED]] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:46   AMTo:   asterisk-users@lists.digium.comSubject: [asterisk-users] ftp   serverWhats the best ftp server to   upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using   BTF server. ___
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Re: [asterisk-users] No Dialtone

2006-10-08 Thread Eddie Johnson Jr

Jay,

I understand.  We can have discussion offline to discuss this further at [EMAIL 
PROTECTED]


Cheers,

Ed
Sent from my BlackBerry® wireless handheld  

-Original Message-
From: "Jay R. Ashworth" <[EMAIL PROTECTED]>
Date: Sun, 8 Oct 2006 14:18:07 
To:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No Dialtone

On Sat, Oct 07, 2006 at 10:10:14PM -0500, Rich Adamson wrote:
> Jay R. Ashworth wrote:
> >On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote:
> >>If you've messed up in connecting telephone lines to the wrong module, 
> >>the ringing voltage sent to a fxs module will destroy it. You would need 
> >>to replace the module.
> >
> >I'm going to stick my neck out here, and opine that any FXS module that
> >would be destroyed by receiving ringing voltage is *incredibly* poorly
> >designed, and very probably wouldn't pass Part 68.  "Shouldn't", certainly.
> 
> Try it and see what happens, and report back. ;)

I didn't say I didn't think current-day hardware *wasn't* designed that
poorly... *just* that it is, in fact, poor design.  (He says, having
see the footage of someone continuing to talk on a 500 desk phone after
someone put a .308 rifle bullet through the network during the call...)

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Re: Real-time and priority "n"

2006-10-08 Thread Brian Capouch


Tony Mountifield wrote:

In article <[EMAIL PROTECTED]>,
Ronald Wiplinger <[EMAIL PROTECTED]> wrote:


Is it exclusive? Either Realtime or priority "n" ???

If so, what is the better way?



I believe 'n' is just a shorthand way of writing "previous line + 1",
and gets converted into an actual number as the dialplan is compiled.
After compilation, the information about whether a line had been given
as 'n' or as a specific number has been lost, as far as I know.



Rows can be added to a database table at any time.  Imagine a series of 
priorities added to a table using nothing more than "n" as a priority 
number beyond the first one.


Now imagine wanting to add a new priority in between any two arbitrary 
entries in the table.  How would you even specify which two lines should 
surround it, when they have no identifying "serial number" associated 
with them?


Unless you were to add a new field, e.g. "priority location identifier," 
or somesuch.  Which does nothing more than move back to the present 
situation.


The extensions.conf parser adds a "real" priority to each line, but in 
Realtime that responsibility falls on the DB maintainer.


B.

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RE: [asterisk-users] ftp server

2006-10-08 Thread Gary Eck

Title: Message



Yes, 
we don't have the proper skillset, either. Even though we are a 
reseller.
 
We 
have standardized on Linux appliances called Snapgear .  Of course, they 
are not integrated into Active directory, so there are lots of things that are 
harder to do.
 
However, since they are not Microsoft or Cisco based, they are not the 
targets of many of the severe attacks on the Internet.
We 
also have problems with installing all the security updates on the scores of 
Small Business Customers we have - they don't want to pay us for installing 
these updates - much less, paying us when a security update has side effects on 
their internal network.
 
So, 
avoiding the whole Microsoft issue is a decent compromise in our 
situation.
 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Sunday, October 08, 2006 1:16 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] ftp server
  
  Unfortunately I 
  disagree with you about ISA, whilst it may cause problems through my lack of 
  skillset from time to time the functionality it introduces and protects my 
  network cant be beat at any price, being built into sbs 2003 is just a 
  bonus.
   
   
  
   
  Cheers,
   
  Dean
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:37 
  PMTo: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp 
  server
   
  
  We also have SBS2003 
  - we initially had ISA running on it, but decided it was not worth the grief 
  at our office.
  
  In the back of my 
  mind, I was wondering if ISA was causing the 
  problem.
  
   
  
  I had used 
  Bulletproof in the past, since it has bandwidth throttling - and it just seems 
  to work fine.
  
  Unfortunately, it 
  does not run as a service - but I found another product that allows it run as 
  a service.
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:30 
PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp 
server
Yep, using sbs 2003 
here.
 

 
Cheers,
 
Dean
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:17 
PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp 
server
 

We had some 
problems with the FTP server on our Windows 2003 server - we worked with it 
quite a bit, but just could not get the Pollycom's to access it. It did work 
with backup up our Switchvox PBX.

 

We use Bulletproof 
FTP server in the Windows world. 
-Original 
  Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:46 
  AMTo: 
  asterisk-users@lists.digium.comSubject: [asterisk-users] ftp 
  server
  Whats the best ftp server to 
  upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using 
  BTF server.
   
   
   
  Cheers,
   
  Dean
   
   
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Re: [asterisk-users] No Dialtone

2006-10-08 Thread Jay R. Ashworth

On Sat, Oct 07, 2006 at 10:10:14PM -0500, Rich Adamson wrote:
> Jay R. Ashworth wrote:
> >On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote:
> >>If you've messed up in connecting telephone lines to the wrong module, 
> >>the ringing voltage sent to a fxs module will destroy it. You would need 
> >>to replace the module.
> >
> >I'm going to stick my neck out here, and opine that any FXS module that
> >would be destroyed by receiving ringing voltage is *incredibly* poorly
> >designed, and very probably wouldn't pass Part 68.  "Shouldn't", certainly.
> 
> Try it and see what happens, and report back. ;)

I didn't say I didn't think current-day hardware *wasn't* designed that
poorly... *just* that it is, in fact, poor design.  (He says, having
see the footage of someone continuing to talk on a 500 desk phone after
someone put a .308 rifle bullet through the network during the call...)

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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RE: [asterisk-users] ftp server

Title: Message








Unfortunately I disagree with you about
ISA, whilst it may cause problems through my lack of skillset from time to time
the functionality it introduces and protects my network cant be beat at any
price, being built into sbs 2003 is just a bonus.

 

 



 

Cheers,

 

Dean

 













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Eck
Sent: Sunday, 8 October 2006 1:37
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] ftp
server



 



We also have SBS2003 - we initially had
ISA running on it, but decided it was not worth the grief at our office.





In the back of my mind, I was wondering if
ISA was causing the problem.





 





I had used Bulletproof in the past, since
it has bandwidth throttling - and it just seems to work fine.





Unfortunately, it does not run as a
service - but I found another product that allows it run as a service.





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Sunday, October 08, 2006
12:30 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] ftp
server

Yep, using sbs 2003 here.

 



 

Cheers,

 

Dean

 













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Eck
Sent: Sunday, 8 October 2006 1:17
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] ftp
server



 



We had some problems with the FTP server
on our Windows 2003 server - we worked with it quite a bit, but just could not
get the Pollycom's to access it. It did work with backup up our Switchvox PBX.





 





We use Bulletproof FTP server in the
Windows world. 





-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dean Collins
Sent: Sunday, October 08, 2006
12:46 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] ftp
server

Whats the best ftp server to upload Polycom phone
cfg’s from? I’m finding it a bit hit and miss using BTF server.

 

 

 

Cheers,

 

Dean

 

 














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RE: [asterisk-users] ftp server

2006-10-08 Thread Gary Eck

Title: Message



We 
also have SBS2003 - we initially had ISA running on it, but decided it was not 
worth the grief at our office.
In the 
back of my mind, I was wondering if ISA was causing the 
problem.
 
I had 
used Bulletproof in the past, since it has bandwidth throttling - and it just 
seems to work fine.
Unfortunately, it does not run as a service - but I found another product 
that allows it run as a service.

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Sunday, October 08, 2006 12:30 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] ftp server
  
  Yep, using sbs 2003 
  here.
   
  
   
  Cheers,
   
  Dean
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:17 
  PMTo: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp 
  server
   
  
  We had some problems 
  with the FTP server on our Windows 2003 server - we worked with it quite a 
  bit, but just could not get the Pollycom's to access it. It did work with 
  backup up our Switchvox PBX.
  
   
  
  We use Bulletproof 
  FTP server in the Windows world. 
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:46 
AMTo: 
asterisk-users@lists.digium.comSubject: [asterisk-users] ftp 
server
Whats the best ftp server to 
upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using BTF 
server.
 
 
 
Cheers,
 
Dean
 
 
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RE: [asterisk-users] ftp server

Title: Message








Yep, using sbs 2003 here.

 



 

Cheers,

 

Dean

 













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Eck
Sent: Sunday, 8 October 2006 1:17
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] ftp
server



 



We had some problems with the FTP server
on our Windows 2003 server - we worked with it quite a bit, but just could not
get the Pollycom's to access it. It did work with backup up our Switchvox PBX.





 





We use Bulletproof FTP server in the
Windows world. 





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Sunday, October 08, 2006
12:46 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] ftp
server

Whats the best ftp server to upload Polycom phone
cfg’s from? I’m finding it a bit hit and miss using BTF server.

 

 

 

Cheers,

 

Dean

 

 










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RE: [asterisk-users] ftp server

2006-10-08 Thread Gary Eck

Title: Message



We had 
some problems with the FTP server on our Windows 2003 server - we worked with it 
quite a bit, but just could not get the Pollycom's to access it. It did work 
with backup up our Switchvox PBX.
 
We use 
Bulletproof FTP server in the Windows world. 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Sunday, October 08, 2006 12:46 AMTo: 
  asterisk-users@lists.digium.comSubject: [asterisk-users] ftp 
  server
  
  Whats the best ftp server to 
  upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using BTF 
  server.
   
   
   
  Cheers,
   
  Dean
   
   
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[asterisk-users] Tellabs and a PRI

2006-10-08 Thread Doug Lytle


Another question,

Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our 
analog lines to a PRI, I thought it would be simple stuff moving the EC 
to the PRI. Changed signaling, made sure that channel 24 wasn't being 
ECd and everything came up. But, I was getting complaints of random echo 
on the PRI. Local echo. Also, we weren’t able to do any kind of modem 
dial-outs (Adit 600 supplying the dialtone). Funny thing is that 
inbound/outbound faxing was working fine.


Removed the Tellabs and turned on the software EC and everything started 
working fine. I'd really like to move back to the Tellabs, so if anybody 
has suggestions, please speak up.


Forgot to mention in the last post, I'm using a Sangoma A102 with the 
current Beta 7 drivers.


Doug


-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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[asterisk-users] PRI issues

2006-10-08 Thread Doug Lytle


Hey everybody,

I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T.  I've 
received several complaints about dropped calls.  Reviewing the archives 
on PRI and dropped calls shows that I should set the resetinterval=never 
in the zapata.conf and restart.  This hasn't helped. 

The dropped calls have to date only been on outbound calls.  Usually 
within 2 to 3 minutes of a call.  The full log shows something about not 
getting a frame and stopping the bridge.


On Saturday I put into place 1.2 Branch and have pri debug setup to log 
to a file.  Is there anything else that I can do to get an idea as to 
what is going on here?


My zapata and zaptel below:

[zaptel]

# Zaptel Configuration File

span=1,1,0,esf,b8zs
defaultzone=us
loadzone=us
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs
fxsks=25-32
fxoks=33-48
defaultzone=us
loadzone=us

[zapata]

[channels]
;
context=default
resetinterval = never
musiconhold=tape

switchtype=national
context=pri
signalling=pri_cpe
group=1
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=-1.0
txgain=-2.0
busydetect=no
pridialplan=unknown
usercallerid=yes
callerid=asreceived
channel => 1-23

I see the following the full log:

Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Executing 
Dial("SIP/4228-082131e8", "ZAP/G1/1xx5800") in new stack

Oct  4 09:09:30 DEBUG[29894] dsp.c: dsp busy pattern set to 0,0
Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Requested transfer 
capability: 0x00 - SPEECH

Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Called G1/xx5800
Oct  4 09:09:30 VERBOSE[29894] logger.c: -- Zap/23-1 is proceeding 
passing it to SIP/4228-082131e8

Oct  4 09:09:32 VERBOSE[29894] logger.c: -- Zap/23-1 is ringing
Oct  4 09:09:37 VERBOSE[29894] logger.c: -- Zap/23-1 answered 
SIP/4228-082131e8
Oct  4 09:11:26 DEBUG[29894] channel.c: Didn't get a frame from channel: 
SIP/4228-082131e8
Oct  4 09:11:26 DEBUG[29894] channel.c: Bridge stops bridging channels 
SIP/4228-082131e8 and Zap/23-1
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: 
ON(1) on Zap/23-1
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Hangup: channel: 23 index = 0, 
normal = 40, callwait = -1, thirdcall = -1
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Not yet hungup...  Calling 
hangup once with icause, and clearing call
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on 
channel 23
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option TDD MODE, value: 
OFF(0) on Zap/23-1
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Updated conferencing on 23, 
with 0 conference users
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: 
OFF(0) on Zap/23-1
Oct  4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on 
channel 23

Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Hungup 'Zap/23-1'
Oct  4 09:11:26 DEBUG[29894] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Oct  4 09:11:26 VERBOSE[29894] logger.c:   == Spawn extension (sip, 
x5800, 5) exited non-zero on 'SIP/4228-082131e8'
Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Executing 
NoOp("SIP/4228-082131e8", "Hungup") in new stack
Oct  4 09:11:26 VERBOSE[29894] logger.c: -- Executing 
Hangup("SIP/4228-082131e8", "") in new stack



-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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[asterisk-users] USA Origination recommended service?

2006-10-08 Thread R.R. Libera


Hello,

I need an advice about USA Origination (only). Who you recommend to me 
for a production environment like a customer care support?


I need:

1) Excellent call quality.
2) Stability.
3) Excellent support.

What will be the best increment schema? 60/15, 60/6, 6/6 ??

Thanks in advance.

R. R. Libera.

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Re: [asterisk-users] Configuration / dialplan problem

2006-10-08 Thread Mark Muffett


I got it to work in the end - by removing the "_" from the front of my
fixed allowed numbers (of course there wasn't any real pattern
matching there at all).

Thanks for the help

Mark

On 03/10/06, Marco Mouta <[EMAIL PROTECTED]> wrote:

If you really want _07. to be tested afterall the above patternmatches, you
must define it in other context and add it as an include for the current
context.

Asterisk first will look for your patternmatches in the current context and
oonly after this will lookup your include context. This way you can avoid
the asterisk "resort"!

pls give some feedback if it helps...


On 10/3/06, Kevin Smith <[EMAIL PROTECTED]> wrote:
> There are a few things to look at.
>
> First off, you have a lot of wildcard testing that is probably throwing
> the dial plan off. For example, you have the following:
>
> exten => _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> exten => _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> exten => _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> exten => _07.,1,Congestion()
>
> If I left it in this order what would happen? From what I understand it
> is nautral to think in that order, but really Asterisk is going to sort
> the extensions something like this:
>
> exten => _07.,1,Congestion()
> exten => _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> exten => _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> exten => _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
>
> So now say you dial 07545865143254/8564, it will go to the Congestion
> application every time.
>
> What I would do is comment out the wildcard searches and see if that
> resolves the problem. If so, try putting all the wildcard tests in an
> include and see if that helps.
>
> Take a look at these to articles as well:
>
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
>
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
>
> Also just out of observation, why all the testing? Seems to me you could
> streamline that code down a bit more. For example, the 01 and 02 tests.
> If you know they are dialing N number of digits, make the test
> _01XX, so you know they have to dial a certain amount of digits
> to be a valid call. Why send a 4 digit number out your trunk if you know
> it isn't going anywhere? If you need to dial '0' then 10 digits, try this:
>
> _01NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> _02NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> _07956X,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) 3
>
> etc.
>
> Hopefully that will help,
>
> Kevin
>
>
> Mark Muffett wrote:
> > I have my extensions.conf set up as follows:
> >
> > exten => _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _01.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _02.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _0800.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _0845.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _0870.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
> > exten => _09.,1,Congestion()
> > exten => _00.,1,Congestion()
> > exten => _07.,1,Congestion()
> >
> > (where nn are actually real digits).
> >
> > I would expect this to let me dial the 07956nn numbers etc while
> > stopping dialing to other 07... numbers, but it seems to stop dialling
> > to any 07... number including the 3 specifically listed.
> >
> > Any ideas?
> >
> > Thanks
> >
> > Mark
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 --
Com os melhores cumprimentos,

Marco Mouta
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Re: [asterisk-users] Xorcom Astribank and 64 bit linux

2006-10-08 Thread Carlos Chavez

On Sun, 8 Oct 2006 10:07:28 +0200, Tzafrir Cohen wrote
> On Sat, Oct 07, 2006 at 04:15:35PM -0500, Carlos Chavez wrote:

> 
> Alternatively, try rebuilding
> http://updates.xorcom.com/trixbox/zaptel-1.2.8-2.xpp.r2223.src.rpm
> 
 I downloaded the file and did a rpmbuild --rebuild on the server but it
did not compile the xpp module.  If I do a rpm -ql of the zaptel-modules I get
this:

[EMAIL PROTECTED] x86_64]# rpm -ql 
zaptel-modules-2.6.9-42.0.2.EL-1.2.8-2.xpp.r2223
/lib/modules/2.6.9-42.0.2.EL/extra/pciradio.ko
/lib/modules/2.6.9-42.0.2.EL/extra/tor2.ko
/lib/modules/2.6.9-42.0.2.EL/extra/torisa.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wcfxo.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wct1xxp.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wct4xxp.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wctdm.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wctdm24xxp.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wcte11xp.ko
/lib/modules/2.6.9-42.0.2.EL/extra/wcusb.ko
/lib/modules/2.6.9-42.0.2.EL/extra/zaptel.ko
/lib/modules/2.6.9-42.0.2.EL/extra/ztd-eth.ko
/lib/modules/2.6.9-42.0.2.EL/extra/ztd-loc.ko
/lib/modules/2.6.9-42.0.2.EL/extra/ztdummy.ko
/lib/modules/2.6.9-42.0.2.EL/extra/ztdummypll.ko
/lib/modules/2.6.9-42.0.2.EL/extra/ztdynamic.ko

 Any extra parameters I need to pass to the rebuild comand?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[asterisk-users] DID is not working (call is not routing)

2006-10-08 Thread Crazy Boy

Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI>    -- IAX2/teliax-2 answered SIP/350-09e3b540    -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk||1") in new stack    -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack     -- Channel will hangup at 2006-10-06 11:27:55 UTC.    -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack    -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack     -- Executing
 Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack    -- Playing 'ss-noservice' (language 'en')    -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack   == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020'    -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack    -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack     -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack    -- Goto (from-sip-external,s,1)    -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack     -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack    -- Channel will hangup at 2006-10-06 11:28:04 UTC.    -- Executing
 Answer("SIP/216.89.79.2-09e1d020", "") in new stack   == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully.  Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. 
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Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-08 Thread Michael Graves



Let me chime on on Astlinux and  my personal experience. I have used Astlinux in the following installations:



1) boot from CF on VIA platform, store config settings on USB key drive

2) boot from CD on P3-800, store config settings on USB key drive

3) boot from CF on Soekris Net4801, store config settings on USB key drive

4) boot from USB key on H-P T5700 thin client platform, store config settings on second partition on USB key drive



In all cases it was diskless, fanless, silent, and utterly reliable. I just can't praise Kristian enough. Astlinux is great.



it doesn't have all the UI fluff of a TrixBox...but you don't need that. I'd rather run on humble (solid state and silent) hardware and shell into it for admin purposes. That said, Astlinux does have a rudimentary web based UI for admin tasks.



I've had these systems in production for two years with now sign of trouble on boot CF or USB keys. If you call my office number, 800 number or via FWD  you're currently be answered by a H-P T5700 thin client with a 1 GHz Transmeta Crusoe processor, 512 MB of RAM and 512 MB of USB key for storage. I saved six of these from being recycled by a large corparate entity.  I'm hoping that the faster CPU lets me do more with G.729 codecs that was possible on a 266 MHZ Soekris Net4801.



Durability in flash based memory is all about minimizing writes. Embedded systems rock! Silence is golden.



Michael Graves





--Original Message Text---

From: Erick Perez

Date: Fri, 6 Oct 2006 11:54:48 -0500



Hi,

Im doing some research for Disk on a Module (DOM) with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting.

 

Does someone have working experience with this?

Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module.

I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles.

 

Is there an utility/section/procedure that can "count/display" the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. 

 

Anyone with field experience?



Thanks,



-- 



Erick Perez

Panama Sistemas

Integradores de Telefonia IP y Soluciones Para Centros de Datos

Panama, Republica de Panama

Cel Panama. +(507) 6694-4780 

 








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[asterisk-users] OH323 Fake ring

2006-10-08 Thread Abdul

Hi All,     I need your urgent help.   i installed OH323 channel and it is working well with samll problem fake ring.  I have my VoIP provider MCI and AT&T when i am routing the call via OH323 from the SIP ATA like Linksys i am getting too much fake ring even some time real RBT is there and also i can hear fake ring.     Could u please guide from which configuration i can disable the fake ring. one thing more funny when i am dialing using SIP channel to these provider i am not getting any fake ring, just we are getting real RBT.     this is my dial option:     AGI Script Executing Application: (Dial) Options: (OH323/MCI/phone-number|350|S(max talk time))     Regards,  Abdul       
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Re: [asterisk-users] astcc help-pleasssssseeee

2006-10-08 Thread Brian Rogan

As I recall, you nee to make sure you run this script with the DeadAGI
command, not just AGI.  This will make sure that the dial command will
return to your script only after it is done.

--Brian

On Sat, Oct 07, 2006 at 10:45:10PM -0700, Ali wrote:
> So what should I do?
> 
> 
> 
> On 10/7/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >
> >On Fri, Oct 06, 2006 at 10:39:18PM -0700, Ali wrote:
> >> Hi,
> >>
> >> I am wondering if astcc has ever worked for someone because it always
> >return
> >> 0 for answeredtime! I tracked every bit of informaion on google and wiki
> >and
> >> finally found out that its because of asterisk returning to dial plan
> >after
> >> executing Dial, so astcc.agi runs through the end without wating for
> >call
> >> completion.
> >>
> >> Am I missing something crazy? please someone give me a hint.
> >>
> >>
> >
> >astcc doesn't use strict. This is perl code I wouldn't like to touch.
> >
> >--
> >Tzafrir Cohen sip:[EMAIL PROTECTED]
> >icq#16849755  iax:[EMAIL PROTECTED]
> >+972-50-7952406  jabber:[EMAIL PROTECTED]
> >[EMAIL PROTECTED] http://www.xorcom.com
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> >

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Re: [asterisk-users] ZAP chanel doesn't reset if external caller hangs up in menu

On Thu, Oct 05, 2006 at 10:38:32AM +1000, Devraj Mukherjee wrote:
> Hello world,
> 
> My asterisk server doesnt seem to disconnect the call if someone
> hangsup say while they are listening to the menu as a result of which
> my phone is engaged forever. Any pointers on fixing this issue?

Which zaptel hardware?

Is the zaptel channel still off-hook? See the last line of 'zap show
channel NNN'

> 
> Thanks
> 
> my extensions.conf
> 
> [incoming]
> exten => s,1,Wait(1)
> exten => s,2,Answer()
> exten => s,3,NoOp(${CALLERIDNUM})
> exten => s,4,Wait(1)
> exten => s,5,Playback(eternity_welcome)
> exten => s,6,Background(eternity_mainmenu)
> exten => s,7,Wait(4)
> exten => s,8,Playback(eternity_loop)
> exten => s,9,Goto(incoming,s,6)
> ; Support
> exten => 1,1,Goto(internal,202,1)
> ; Server and support
> exten => 2,1,Goto(internal,102,1)
> ; LiveCD support
> exten => 3,1,Goto(internal,101,1)
> ; Reception
> exten => 0,1,Goto(internal,102,1)
> exten => s,10,Hangup()

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Optus PRI via DSL

2006-10-08 Thread Paul Hales


I have seen an Optus SHDSL box set up incorrectly before - and the tech
re-visited and set it up correctly within hours of being informed.

PaulH

On Fri, 2006-10-06 at 12:17 +1000, [EMAIL PROTECTED] wrote:
> There was a bit of traffic on this list a while ago regarding OPTUS
> multi line that comes in via a DSL box, and I am hoping some of those
> people are still hanging around, and solved their problems.
> 
> We apparently have a 14B channel service with optus.
> 
> I have been trying to configure Asterisk, with a TE110p card with no
> luck at this stage.
> 
> I have consoled onto the DSL box and have found the following:
> The box is an ADTRAN 408582526 SHDSL box
> The operation mode is NTU G.703 Unframed (Indep. Clk)
> 
> 
> Now when I plug in an Rj-45 from the DSL box to the TE110p I get
> nothing, and so I have made an E1 crossover, and I now get a combination
> of green lights. One on the back of the TE110p, and the G.703 light on
> the DSL modem comes on.
> 
> This would suggest to me that the physical link is working fine.
> 
> The following comes up at boot:
> TE110P: Setting up global serial parameters for E1 FALC V1.2
> TE110P: Successfully initialized serial bus for card
> Found a Wildcard: Digium Wildcard TE110P T1/E1
> Registered tone zone 1 (Australia)
> TE110P: Span configured for CCS/HDB3/CRC4
> 
> 
> my /etc/zaptel.conf is as follows:
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-14
> dchan=16
> unused=15,17-31
> 
> loadzone = au
> defaultzone = au
> 
> 
> When I try and dial our inbound number, I just get a busy signal.
> 
> The fact that NTU G.703 is operating in Unframed mode on the DSL
> looks to be the problem to me, however I am in over my head a little
> with this stuff.
> 
> Unfortunately I do not have the management password to login and change
> the framing on the HDSL modem.
> 
> Would I be correct to say if I use clear channel on the TE110p (ie not
> use framing) it would work, and how do I set this on the card?
> 
> Anyone have any ideas of where I should go from here?
> 
> Thanks in advance
> 
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[asterisk-users] Sun Cluster and Asterisk

2006-10-08 Thread Giedrius Augys

Hi, I'm very interesting in high availability sollutions for ASTERISK. And I know that big companies are using SOLARIS (and I think SUN CLUSTER) for their VoIP gateways. Solaris and Sun Cluster are both free, of course you must pay if you want support. But I have read in FAQ this:
Can i install Asterisk on a beowulf cluster? A
cluster can't migrate threads that use shared memory. Asterisk uses
that kind of threads.So no, Asterisk wouldn't work on a cluster. (It
might be helpful to know whether anyone has a working load-balanced
Asterisk configuration where multiple systems can share the load of an
Asterisk environment (IAX2, not SIP) and whether this environment would
fail over nicely in the event of downtime!)
So my question is: has somebody built asterisk in solaris(from source ) and using Sun Cluster? 
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Re: [asterisk-users] ftp server



On 08/10/2006, at 3:00 PM, Dean Collins wrote:

Whats the best ftp server to upload Polycom phone cfg’s from? I’m  
finding it a bit hit and miss using BTF server.


I'm using vsftpd quite successfully on several Asterisk boxes with  
Polycom IP501 phones. Though, I'm now considering switching to HTTP  
provisioning so that I can actually dynamically create Polycom  
configurations from a MySQL database.


At the moment, its all vaporware, but its a nice idea. :)

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Re: [asterisk-users] How can i store PAP2 or any device config in Asterisk

2006-10-08 Thread ram

Hi
 
thanks for the quick reply
i will look on that
 
thanks
 
Ram 
On 10/8/06, Yair Hakak <[EMAIL PROTECTED]> wrote:

aterisk does not do this, you need a provisioning server. google for pap2 and tftp.
-yair 


On 10/8/06, ram < [EMAIL PROTECTED]> wrote:
 



Hi all
 
I have installed asterisk
 
when any of the user device made on, it should contact Asterisk
and download the config
 
how can i asterisk does this job, does asterisk does
 
or i should have any other server to meet my requirement
 
Ram___--Bandwidth and Colocation provided by 
Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel 
israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] -- 
Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israelisrael: (972) 54 5491266usa: (212) 202 2340
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Re: [asterisk-users] G729 Licence Consumption Problem

2006-10-08 Thread Thomas Kenyon


Alvaro Parres wrote:

Hi List:

I have the next diagram:
GSM   G729
G729  
 IdeFisk -- Asterisk A - 
[INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )


   The user at IdeFisk Login as Agents on Asterisk B
at this moment we have the next Licence Use:
  A) 1/1
  B) 1/0

   When a Call from the QUEUE on Asterisk B is Bridge to the Agent I 
have the next Use:


 A) 1/1
 B) 1/3

Any one can explain me this ?, why the incress of licence consumptions.

Thanks.

Are you recording the call? There may be a separate process decoding the 
call (each way) to make a recording.

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[asterisk-users] Transfer app and DTMF via SIP info

2006-10-08 Thread Michael Konietzny


Hello asterisk-users,

I'm currently investigating a problem related to the Transfer app and
DTMF tones via SipInfo.
My setup depends on:

Asterisk  1.2.10
Zaptel 1.2.8
libpri 1.2.3
Elmeg IP 290 (snom190)
Wildcard TE400 (E1)

The following dialplan is given:

exten => 555, 1, Transfer(554);

exten => 554, 1,Dial (SIP/tel3, 10, tT);
exten => 554, 2,Dial (Zap/g1/017123123123, 10, tT);
exten => 554, 3,Hangup();

If I dial 555 on my SIP phone it transfers to 554 and connecting me to
that zap channel.
Arriving there I'm not able to type ANY DTMF tones.

If the Transfer is skipped the DTMF tones are available. I've included
the SIP debugs to help you track the problem.

Greetings and many thanks in advance,

Michael Konietzny

-- Executing Transfer("SIP/tel2-b721ef28", "554") in new stack
Reliably Transmitting (no NAT) to 192.168.97.21:2054:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 
192.168.97.21:2054;branch=z9hG4bK-gu4c0f6c0cim;rport;received=192.168.97.21
From: "tel2" ;tag=r7pzlq4bdy
To: ;tag=as21b6ba81
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: Transfer 
Content-Length: 0

...

-- Called tel3
-- SIP/tel3-082c99c8 is ringing
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21
From: "tel2" ;tag=lt4rnm3do0
To: sip:[EMAIL PROTECTED];tag=as20294491
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 31556 31556 IN IP4 192.168.97.11
s=session
c=IN IP4 192.168.97.11
t=0 0
m=audio 18426 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 telephone-event/8000
a=fmtp:0 0-16
a=silenceSupp:off - - - -

Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport
From: "tel2" ;tag=lt4rnm3do0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 2 CANCEL
Max-Forwards: 70
Contact: 
Content-Length: 0

... 

-- Called g1/017123123123
We're at 192.168.97.11 port 18426
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.97.21:2054:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21
From: "tel2" ;tag=lt4rnm3do0
To: sip:[EMAIL PROTECTED];tag=as20294491
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 31556 31556 IN IP4 192.168.97.11
s=session
c=IN IP4 192.168.97.11
t=0 0
m=audio 18426 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 telephone-event/8000
a=fmtp:0 0-16
a=silenceSupp:off - - - -



-- Hungup 'Zap/1-1'
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-7zpvewacvy6j;rport
From: "tel2" ;tag=dtndk3lw7m
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: 
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1728010931 1728010931 IN IP4 192.168.97.21
s=call
c=IN IP4 192.168.97.21
t=0 0
m=audio 62868 RTP/AVP 8 0 3 9 18 4
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=ptime:20
a=sendrecv

... 

INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-qgjwhifwkg39;rport
From: "tel2" ;tag=dtndk3lw7m
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Max-Forwards: 70
Contact: 
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest 
username="tel2",realm="asterisk",nonce="61364bf6",uri="sip:[EMAIL 
PROTECTED];user=phone",response="5140f1d5f042256b8daf901b18c603af",algorithm=md5
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1728010931 1728010931 IN IP4 192.168.97.21
s=call
c=IN IP4 192.168.97.21
t=0 0
m=audio 62868 RTP/AVP 8 0 3 9 18 4
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=ptime:20
a=sendrecv

-- Called tel3
wum97011*CLI>
<-- SIP read from 192.168.97.21:2054:
SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-xpi4mpqz7oo7;rport
From: ;tag=0vtl8gobz9
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 2 SUBSCRIBE
Max-Forwards: 70
Contact: 
Event: di

Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability

2006-10-08 Thread Brian Candler

On Fri, Oct 06, 2006 at 05:10:34PM -0500, Erick Perez wrote:
>Example: if I setup system XYZ with asterisk, then load this magical
>utility/procedure that counts how many writes the filesystem has done
>to / or to /,/tmp,/var and after 24 hours the utility/procedure says:
>10thousand writes, then, I will do

It's not so much magic. You're looking for "iostat", see
http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/admin-primer/s1-resource-rhlspec.html
http://www.redhat.com/magazine/011sep05/features/tools/

I'm no expert here, but for flash drives it might not be quite so simple.
You may need to think about how many times the *same* block is overwritten.

If you use a filesystem which does "wear levelling" (such as jffs2), it will
spread these operations across the whole device. And maybe your flash device
does wear levelling of its own, in which case you don't have to worry.

http://en.wikipedia.org/wiki/Flash_memory
http://en.wikipedia.org/wiki/Wear_levelling

Then you just need to check whether your device spec for 2 million R/W
cycles is the value per block, or the total value spread across the whole
device. Since the latter gives the higher number, then that's probably what
the manufacturer has quoted.

HTH,

Brian.
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Fwd: [asterisk-users] How can i store PAP2 or any device config in Asterisk

2006-10-08 Thread Yair Hakak

aterisk does not do this, you need a provisioning server. google for pap2 and tftp.
-yair 

On 10/8/06, ram <
[EMAIL PROTECTED]> wrote: 


Hi all
 
I have installed asterisk
 
when any of the user device made on, it should contact Asterisk
and download the config
 
how can i asterisk does this job, does asterisk does
 
or i should have any other server to meet my requirement
 
Ram___--Bandwidth and Colocation provided by 
Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel 
israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] -- Yair Hakak
-Yair Hakak, CEOGo Telecom, Ltd., Israelisrael: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] 
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Re: [asterisk-users] Xorcom Astribank and 64 bit linux

On Sat, Oct 07, 2006 at 04:15:35PM -0500, Carlos Chavez wrote:
>  DOes anyone know if you can use an Astribank usb channelbank with a 64
> bit linux distribution like CentOs?  I saw a note that the driver is only
> built when you have an i386 processor and a kernel>= 2.6.10 

Actually this has been changed to "and also if arch is amd64 and kernel
version >= 2.6.10" . 

> but it does not
> mention if you can build it manually for other architecture/kernel.

The Astribank drivers work just fine on my workstation:

Linux frenkel 2.6.15-1-amd64-k8 #2 Tue Mar 7 20:57:25 UTC 2006 x86_64 GNU/Linux

Should work fine on amd64 in general.

There is one expected build problem with recent RHEL kernel which I have
not been able to work around in a decent manner:

http://lists.digium.com/pipermail/asterisk-dev/2006-September/023329.html

Thus you'll probably need manual patching of the tarball.

Alternatively, try rebuilding
http://updates.xorcom.com/trixbox/zaptel-1.2.8-2.xpp.r2223.src.rpm

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] How can i store PAP2 or any device config in Asterisk

2006-10-08 Thread ram

Hi all
 
I have installed asterisk
 
when any of the user device made on, it should contact Asterisk
and download the config
 
how can i asterisk does this job, does asterisk does
 
or i should have any other server to meet my requirement
 
Ram
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[asterisk-users] RE: ftp server