from:"Al Baker"





You mean running , 400 Calls on 1 BOX ?
Even if you COULD do it, the gods of TELCO would have you burn in hell
for stacking that much critical traffic on ONE Intel, non - high
availability box

Jerry Geis wrote:

  Assuming you have a Quad core machine, at least 4 GIG ram,
will a machine like this handle 4 Quad T1 cards?

is that advisable?

What about running AGI's on such a machine.
Will the machine handle starting/stopping all those AGI's?

Thanks,

Jerry

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Re: [asterisk-users] Custom Filename for Incoming Agent Calls

Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL 
to that call ??
Thanks

lenz wrote:
 Hi Ricardo,
 Try this:

 exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
 exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
 exten = s,13,queue(q-pa|t|||)

 The TRANSFER_CONTEXT is used for transfers. If you need the filename  
 inherited, add a double underscore before it.
 Thanks
 l.



 In data Wed, 30 Jul 2008 23:09:11 +0200, Ricardo Melendez  
 [EMAIL PROTECTED] ha scritto:

   
 Hi, to all, I have configured 3  Inbound/outbound agents queues,  I  
 record
 Outgoing calls with custom filename like
 outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm

 but I need to record Incoming calls and asterisk by default add 13 digits
 number to inbound recordings  like Agent-001-1298375678-890.gsm, how I  
 can
 customize this filename recordings?


 Thanks in advance.


 Ricardo Melendez


 



   

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Re: [asterisk-users] HI ~ good friend,

I must disagree.
Dimensioning of Asterisk is a very sorely lacking area and is one of 
the main area CISCO
and such eats its lunch. There simply no a base of solid metric that 
allow for true provisioning .
Yes, there are INVALUABLE anecdotal reports from people who have been 
kind, and sharing of their
experiences and for which are all very very grateful.
BUT
That that just is not the same as as solid, vendor based Metrics.
Can you imagine calling and asking DISCO, What do I need for 400 calls 
an their answer is
Here please go read these mostly outdated anecdotal reports and call 
back with your order
Sorry. I love *, but this  area of it is not where it needs to be.

Dean Collins wrote:

 Hi welcome to the asterisk community.

  

 The answer you want are here; 
 http://www.voip-info.org/wiki/view/Asterisk+dimensioning

  

 The short answer is; Pretty much yes, depending on hardware and 
 horizontal scaling with multiple servers sharing the load.

  


 Cheers,

 Dean

 

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *???
 *Sent:* Friday, 1 August 2008 9:43 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] HI ~ good friend,

  

 hi ~ nice to meet you, i just join here, today,

  

 i am a student, and i am very interesting in asterisk.

  

 and i have a IP-PBX server, made by me with my friend,

  

 while when i studying, i have a question,

  

 is there any limit users for asterisk?

  

 ex) registed users number is 1000 or 1 or 10 like that, is 
 that possible?

  

 and how about the concurrent calls? 1000 concurrent calls is possible? 
 or 2000 concurrent calls?

  

 my PBX server's user is just less then 15, almost my friends,

  

 so, i can't test, over 10 users and 1000 concurrent calls,

  

 please tell me, it is possible or not?

  

 thanks your permission to join there,




 

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Re: [asterisk-users] Custom Filename for Incoming Agent Calls

Wow - Thanks a bunch. Likely save me about 12 hours of struggle

Tilghman Lesher wrote:
 On Friday 01 August 2008 14:16:42 Al Baker wrote:
   
 lenz wrote:
 
 Hi Ricardo,
 Try this:

 exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
 exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
 exten = s,13,queue(q-pa|t|||)

 The TRANSFER_CONTEXT is used for transfers. If you need the filename
 inherited, add a double underscore before it.
   
 Could you clarify what you mean as inherited
 In The dial plan for a given call I thought All variable were GLOBAL
 to that call ??
 Thanks
 

 No, variables are global to the CHANNEL.  Calls are bridges between two
 channels.  Variables are not transferred to a dialled (slave) channel, unless
 you set up inheritance, as noted above.

   

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Re: [asterisk-users] announcement server using asterisk

2008-07-28 Thread Al Baker

Quote 

Recently I discovered a cool new site called Google.
They have lots of information about ISDN cards.  :-P

Grüße,
Philipp Kempgen


Yes - There is also a lot of bogus, incorrect, crap.
His question was  fair, on-topic, politely asked and as such hardly deserves to 
be 
made fun off


Dean Collins wrote:
 Lol crackup.

 Having said that here is some help.

 Don't even think of using a laptop that's just dumb.
 Next - check out www.voip-info.org you'll find what you need there.


 Regards,

 Dean Collins

 +1-212-203-4357 (Direct) 
 +61-2-9016-5642 (Sydney in-dial)
 http://www.Cognation.net

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
 Sent: Saturday, 26 July 2008 9:57 AM
 To: Asterisk Users
 Subject: Re: [asterisk-users] announcement server using asterisk

 ballamudi madhulika schrieb:

   
 Can I use Asterisk as an announcement server. We want to build announcement
 server with ISDN PRI card terminating on our server and announcement being
 fed on the incoming calls.
 

 Yes.

   
 Also is there any ISDN card available for Laptop.
 

 Recently I discovered a cool new site called Google.
 They have lots of information about ISDN cards.  :-P

 Grüße,
 Philipp Kempgen
   

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Re: [asterisk-users] Cisco vs Asterisk

2008-07-25 Thread Al Baker

Quote

Yet amazingly (if this is, indeed, a source of amazement for you), CCM 
and other Cisco software can be just as buggy as anything OSS, if not 
worse. 

This is simply NOT TRUE and shows a complete lack of understanding of modern 
software development.
CISCO software is developed in a CMM environment.
It has a formal test methodology and uses Automated Testing on EACH new release 
to ensure that 100% of the software that functioned in the Last Release, 
actually works in this release.
Further, there is mandatory soak-testing  for all new software.
Sorry, anyone who wants to compare Professional TELCO GRADE software 
development with Open Source is just Completely and Totally freakin clueless.


Alex Balashov wrote:
 T G wrote:
   
 I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and 
 Telepresence systems I have two IP patents for the VoiP Lite protocols 
 and have been designing and building OSS IPBXs for companies including 
 Google going back to 2001.
  
 I'm not mentioning any of that to be jerk I mentioned it to say I'm as 
 qualified as anyone to to compare the CCM and OSS servers.
  
 The only fair way to compare the two is a list of weights features, for 
 example if cost is your biggest feature then OSS is better, if support 
 is your biggest feature than Cisco wins.
  
 When a customer is comparing the costly (TCO) and best supported systems 
 in the world with hundreds of thousands installed systems for the large 
 global companies on the planted backed by 54,000 employees and over $25b 
 in the bank vs, a FREE system with one layer of support maybe two layers 
 of support, the features don't even come in the evaluation in my opinion.
  
 I once asked a manager why did you buy the CCM and he said no one ever 
 got fired for buying Cisco if anything wrong, If push the OSS and it 
 goes I could loose my job.
  
 I would get a list of the important features, because there is no answer 
 to your question of which is better.
 

 Yet amazingly (if this is, indeed, a source of amazement for you), CCM 
 and other Cisco software can be just as buggy as anything OSS, if not 
 worse.  Depending on how critical the bugs or other support exigencies, 
 the TCO can be driven way up.

 Except with the OSS community, you report the bug, and usually get a 
 quick fix - even if it's a significant issue for you, not necessarily 
 most of the installed base.  If by chance that proves not to be the 
 case, the source code is available, and you can fix it yourself.

 With Cisco, you pay for expensive support and get to file some complaint 
 with the TAC.  Yay.

 There are many, many angles from which onec an look at this in one's TCO 
 / OPEX formula.

   

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Re: [asterisk-users] Cisco Call Manager to Asterisk conversion

2008-07-25 Thread Al Baker

Quote I need to replace a cisco call manager with an asterisk box.
WHY ?
You want your TELCO to be LESS Reliable with LESS SUPPORT 

Grygoriy Dobrovolskyy wrote:
 Search someone in local area, remote configuration of server is 
 possible but configuring the phones is more difficult, you need 
 someone to load firmwares, ect

 2008/7/24 Chad Whitten [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]:

 I need to replace a cisco call manager with an asterisk box.  Phones
 are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's
 have to use the skinny/sccp driver.  Its been quite awhile since I did
 anything with asterisk, so I am looking for some assistance with the
 configuration and am willing to pay.  Its a basic setup, 30+ phones,
 incoming lines via PRI, 1 dial plan for incoming and outgoing -
 nothing fancy there, voicemail for each phone and DID number for each
 phone.

 --
 Chad Whitten
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

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Re: [asterisk-users] Acceptance testing of a new PRI

2008-07-25 Thread Al Baker

Quote

seems like a dial-by-span syntax.
What is Dial-by-span ?

I have looked and cannot seem to fund that term.
More likely a comment on my ability to find it than on it obscurity


Tzafrir Cohen wrote:
 On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
   
 On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
 
 What's wrong with plain old Zap/NN ?

 [test]
 exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})

 Now call 6chan_numnumber-to-dial in context test.
   
 As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
 the argument to Dial, I get CHANUNAVAIL.
 

 Zap/01-1 ??? How come?

 Zap/01 is valid and equivalent to Zap/1 .

   
 So I guess I need finally to end up with 

 exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
 exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
 

 Err.. that's not mine. It seems like a dial-by-span syntax.

 Just remove the '-1' .

   
 exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
 exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
 exten = _88XX1NXXNXX,5,Hangup

 exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
 exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
 exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
 exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
 exten = _880X1NXXNXX,5,Hangup

 Which I just retested and it works.

 Now to figure out how to do it across IAX channels from one server to
 another.

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

   Those who cast the vote decide nothing.
   Those who count the vote decide everything.
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Re: [asterisk-users] How to monitor Asterisk logs ?

2008-07-16 Thread Al Baker


I think this is another area DIGIUM has failed to address in any 
meaningful way.
If they TRULY  see themselves as a TELCO replacements for large shop 
they REALLY need to step up to
proving INFO, WARN, ERROR messaging in a unified reliable manner. Such 
as a SNMP messaging ability for all
INFO, ERROR, and WARN level messages.
The very ideal of having to parse a log file for error messages whose 
form and meaning  may be added to , deleted, or changed
by the vendor at any time is just reasonable
Anthony Francis wrote:
 perl script.

 Olivier wrote:
   
 Hi,

 How can I be notified anytime a given warning message appears in 
 Asterisk logs ?

 I've got a running system that produces cause 34 warnings (Unable 
 to create channel of type 'Zap' (cause 34 - Circuit/channel 
 congestion)) once or twice a week.
 I would like to like to be notified (by email, phone, ...) anytime 
 such warning message occurs in log file.

 I was thinking of using logwatch but wondered if anything better exists.
 Any advice ?

 Regards
 

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Re: [asterisk-users] Asterisk as an IVR solution

SO does that mean that if he used BACKGROUND is a SubRoutine  he would
get the correct or desired action , from his point of view? It would 
jump to the 1 Extension in the SUBROUTINE ?

Tilghman Lesher wrote:
 On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
   
 It's a known problem.

 If you call Background() in a macro, then Asterisk will look for the
 extensions to jump to in the CALLING Macro/context and NOT the Macro that
 the Background() app was called in.
 

 I wouldn't call it a known problem.  It works precisely as it was designed to
 work.  It may not work the way that you want it to, but it works like a Macro:
 an independent set of instructions, with substitution, that acts as if it were
 invoked inline with the calling location.  That is why Background will match
 in the context of the calling location: it acts like it never left that
 original context (and, in a way, it really didn't).

 Subroutines are a different beast, and they are available with the Gosub/
 Return set of routines in app_stack.so.

   

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Re: [asterisk-users] Diagnosing dropped calls...

Quote Seriously though, if your business lives and dies by the phone 
system,
  get T1 with SIP from your provider directly 

If your business lives and dies, get that regular, boring, RELIABLE, TDM-T1.
SIP/VOIP/Whatever - Cool fun, great when it works
TDM-T1 - Unsurpassed reliabilty

Steve Totaro wrote:
 Unfounded rumors say that ABE doesn't come with app_rnddropcall ;-]


 On Fri, Jul 11, 2008 at 12:40 PM, Carlos Chavez 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:

 The other thing that baffles me about this setup is that it only seems
 to happen to people who are connected to the internal network in the
 office. They have about 30 remote users that have not reported this
 same problem, their issue is usually bandwidth related from their home
 connection.

 We have checked the internal network several times and there is
 not any
 obvious problem (apart from the dropped calls). They use high end
 Cisco
 switches and they were just audited to make sure there were no
 configuration errors.

 All the internal phones are Aastra (most are 9133i and some others
 53i).

 On Thu, 2008-07-10 at 20:55 -0400, Steve Totaro wrote:
  Try dropping the IAX2 and only use SIP. Don't ask why? Just give
  it a try and see if things improve for you.
 
  Also when you assume, you make and ass out of you and me (just a
  little joke, get it? ass-u-me.)
 
  You could be hitting an overloaded router or whatever along the way,
  10mbs fiber does not mean low latency or lost packets.
 
  Seriously though, if your business lives and dies by the phone
 system,
  get T1 with SIP from your provider directly (point to point)
 with G729
  or just get a real ISDN or POTS lines.
 
  And then you will still have dropped calls depending on your
 volume
  and how vocal your users are. Usually, once they perceive a problem,
  then even if the other side of the call is on a cell and the cell
  drops the call, you will get a complaint. The only way to track
 those
  down are on a case by case basis with ANI II codes 61-63
  http://www.nanpa.com/number_resource_info/ani_ii_assignments.html
 
  Thanks,
  Steve Totaro
 
  On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
  My customer has a 10mpbs fiber connection to the
  Internet so we have
  always assumed that the connection is not really a problem.
  We will
  look into it. Thank you.
 
 
 
 
 
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Re: [asterisk-users] Asterisk as an IVR solution

Could you clarify how you end up with 1.4 Backport ?
If you go to DIGIUM and download 1.4 do you have a backport 1.4 or is 
there
a super-secret-non-more-secret-archive one would get it from ?
I have never really understood this.
Thank You

Tilghman Lesher wrote:
 On Friday 11 July 2008 12:07:37 Douglas Garstang wrote:
   
 A subroutine with arguments?
 

 In 1.6, yes, or in the 1.4 backport, yes.

   

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Re: [asterisk-users] Asterisk as an IVR solution



Douglas Garstang wrote:
 Well, a macro is the closest thing the dial plan has to a subroutine, 
 and without that, we might as well be programming in Assembler (no 
 subroutines, local variables, lots of goto's... sound familiar?).

 Doug.

 - Original Message 
 From: Tilghman Lesher [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, July 11, 2008 7:20:40 AM
 Subject: Re: [asterisk-users] Asterisk as an IVR solution

 On Friday 11 July 2008 01:28:34 Douglas Garstang wrote:
  Well I can tell you that it makes a difficult programming 
 environment, just
  a little more difficult. It means I can't implement a menu as a single
  reusable piece of code inside a macro.

 That's the point.  A Macro is NOT a subroutine.  It's like saying that you
 can't effectively hammer a nail with a screwdriver, and therefore you 
 think
 the screwdriver has a known problem.  There's nothing wrong with the
 screwdriver; it simply is the wrong tool for the job.

I must somewhat disagree with you on this.
1) A MACRO could reasonably viewed as the Current Context, so if the 
jumping/branching from extension to extension that takes place in other 
contexts, it would if fact be quite reasonable and expected that this 
would happen in a MACRO.
2) As SUBROUTINES did not come standard until 1.6, it might be 
reasonably stated that no appropriate tool existed until 1.6,
and since good programming practice uses subroutines, and a MACRO did 
not work like subroutine, even though it LOOKS like one, people are not 
fully happy that the closest tool they had, did not do the job

Just a thought , no flame intended or implied.

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Re: [asterisk-users] Asterisk as an IVR solution

Thank You - clears up a LOT I did not fully grasp

Tilghman Lesher wrote:
 On Friday 11 July 2008 01:05:22 Al Baker wrote:
   
 Tilghman Lesher wrote:
 
 On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
   
 It's a known problem.

 If you call Background() in a macro, then Asterisk will look for the
 extensions to jump to in the CALLING Macro/context and NOT the Macro
 that the Background() app was called in.
 
 I wouldn't call it a known problem.  It works precisely as it was
 designed to work.  It may not work the way that you want it to, but it
 works like a Macro: an independent set of instructions, with
 substitution, that acts as if it were invoked inline with the calling
 location.  That is why Background will match in the context of the
 calling location: it acts like it never left that original context (and,
 in a way, it really didn't).

 Subroutines are a different beast, and they are available with the Gosub/
 Return set of routines in app_stack.so.
   
 SO does that mean that if he used BACKGROUND is a SubRoutine  he would
 get the correct or desired action , from his point of view? It would
 jump to the 1 Extension in the SUBROUTINE ?
 

 Yes, if he used Background within a Gosub, it would behave the way that he
 expects.

   

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Re: [asterisk-users] Asterisk as an IVR solution

2008-07-10 Thread Al Baker

Why can't you call Background() from a MACRO ?
Isn't is just an Application like any other ?
Curious minds want to know !

Quote There's also the fact that you can't
  call Backgound() in a macro,

Douglas Garstang wrote:
 Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the 
 input in seconds. If you try and use 0, it seems to drop back to a 
 default of 5s.

 - Original Message 
 From: MFH [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, July 10, 2008 12:37:31 PM
 Subject: Re: [asterisk-users] Asterisk as an IVR solution

 From what I can tell Read allows for a floating point input which uses
 ast_waitfordigit that accepts milliseconds as input.

 Douglas Garstang wrote:
  Admittedly I have not used the ExternalIVR app. Is it any good?
 
  I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure,
  it can do it, but boy it is UGLY. There's also the fact that you can't
  call Backgound() in a macro, which forces you to use Read() which
  won't accept a timeout of 1s. There's no DTMF background detection
  while playing SayDigits so you have to roll your own by calling an
  external AGI and concatenating sound files. Yuck. By the time you code
  in logic for handling timeouts and incorrect responses to menu's with
  all the gotos and what-not, it turns into a god aweful mess.
 
  Sure, you can do it.
 
  Doug.
 
 
 
  - Original Message 
  From: Steve Totaro [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com 
 mailto:asterisk-users@lists.digium.com
  Sent: Thursday, July 10, 2008 10:37:55 AM
  Subject: Re: [asterisk-users] Asterisk as an IVR solution
 
 
 
  On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
 Hi.
 
 We are building an application that will provide users with the
 ability to call in and report an absence. The caller will have to
 validate themselves and the call tree will be dynamic, based on
 data in a MySQL database. We will have many customers, each
 calling a separate phone number, each having a different call
 tree. New customers will be added regularly and we do not want a
 solution that requires extensive programming each time (the call
 trees are different in subtle ways from each other).
 
 Is Asterisk a great solution for this? If not do you know what
 would? If so, we need someone to help us set it up, can you
 suggest someone?
 
 Thanks in advance. Best.
 
 Mark
 
 
  Asterisk certainly is a great solution for this.  If you find you need
  or want extra flexibility,  the external IVR app. 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
 
  Thanks,
  Steve Totaro
 
  
 
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Re: [asterisk-users] Asterisk as an IVR solution

2008-07-10 Thread Al Baker

Yes , you could easily do this with asterisk.
If you have formal specs for this project, I would be interested in exactly
what you are trying to do. Email me off-line.

Steve Totaro wrote:


 On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 Hi.

 We are building an application that will provide users with the
 ability to call in and report an absence. The caller will have to
 validate themselves and the call tree will be dynamic, based on
 data in a MySQL database. We will have many customers, each
 calling a separate phone number, each having a different call
 tree. New customers will be added regularly and we do not want a
 solution that requires extensive programming each time (the call
 trees are different in subtle ways from each other).

 Is Asterisk a great solution for this? If not do you know what
 would? If so, we need someone to help us set it up, can you
 suggest someone?

 Thanks in advance. Best.

 Mark


 Asterisk certainly is a great solution for this.  If you find you need 
 or want extra flexibility,  the external IVR app.  
 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR

 Thanks,
 Steve Totaro
 

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Re: [asterisk-users] Loose connection with MySql.

2008-06-24 Thread Al Baker

errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the 
database ?!?
WTF
Since the CDRs are the literal Cash and Life Blood of many application 
why the heck would it NOT do this as part of its minimal basic operation ???

If it Doesn't do this for CDRs does it NOT do it for RealTime ??
If not, one could it up,screwed,blued and tatoed
Is this functionality or lack there of documented anyplace ???

Michiel van Baak wrote:
 On 09:54, Tue 24 Jun 08, Catalin S. wrote:
   
 Hello,
 I configured asterisk to use mysql for CDR. Well when i check from time to
 time I realize
 that asterisk loose connection with mysql (i use phpmyadmin and i watch the
 processes).
 Can anybody tell me how can i solve that problem? I want to have all cdr
 statistics logged in mysql,
 is very important for billing.

 Thank you for support.
 

 Use cdr_adaptive_odbc backport for 1.4.
 That one does a check if the connection is still working, and if not it
 will reconnect.
   

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Re: [asterisk-users] Dial Command Option D Early Bridged

2008-06-24 Thread Al Baker

How do other applications, such a the automated dialers from telemarketers,
reliably detect when the call has been answered ?
I thought this sort of basic functionality that had been around for 
quite awhile.


Jared Smith wrote:
 On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote:
   
 However, in my experience, the timing the call get bridged is not
 consistance,
 

 Do you happen to be calling out over an analog phone line?  In the case
 of dialing out an analog line, we have no easy way of knowing when the
 far-end has answered the call, so the call is considered answered at the
 time the call is dialed.

   

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Re: [asterisk-users] MeetMe Limits

2008-06-08 Thread Al Baker

The 2 big questions are:
-Are all participants using QoS end to end ?

-Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
the work on the * box goes up and can be a problem.

Sam wrote:
 I am thinking about using my asterisk server to host a conference with 
 about 12 other people from around the USA.  Bandwidth issues aside, will 
 this work or will all the different latencies cause issues?  Yea I know, 
 I could just try it and find out but it is going to take alot of time 
 to get everyones schedule to line up, I don't want to go through the 
 trouble if I will just be disappointed.

 Thanks,

 Sam

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Re: [asterisk-users] Mysql and extensions.conf

2008-06-04 Thread Al Baker

yes - but what would REALLY BE GOOD is if func_odbc
allowed Muli-stepped SQL. Since that is the ONLY way to execute a 
TRANSACTION
How they thought it was a Good Idea to hamstring func_odbc like they 
did is beyond me.

Tilghman Lesher wrote:
 On Monday 02 June 2008 05:48, Atis Lezdins wrote:
   
 You can use func_realtime in dialplan, that will be much faster as it
 doesn't create separate process (as AGI does), and uses internal
 asterisk connection pool, so no extra code in dialplan.

 http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime
 

 That assumes that he's using a realtime table.  From the OP's description,
 it sounded like he wanted to query a column of an arbitrary table.  Another
 solution, in addition to the MYSQL app, would be func_odbc:

 func_odbc.conf:
 [FOO]
 dsn=mysql-asterisk
 read=SELECT status FROM foo WHERE id='${ARG1}'

 extensions.conf:
 GotoIf($[0${ODBC_FOO(123)}  0]?open:closed)

   

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Re: [asterisk-users] Asterisk just stops working...

2008-06-04 Thread Al Baker

No - I just would like to suggest that if you provide a solution in a
more clear English manner, more people can benefit from you knowledge
Which I assume is why you posted it in the first place.

Jay R. Ashworth wrote:
 On Thu, May 29, 2008 at 04:24:57AM -0400, Al Baker wrote:
   
 Quote

 THen, fire up under the debugger. When you're all locked up, use ^C to
 
 halt and leave the debugger in command, and do the thread apply all bt
 thing. That should be revealing.
   
 If I may suggest , what would REALLY be 'Revealing' is if you could be 
 just a bit more clear in your explanation and about 900% LESS in the 
 techno babble.
 While the thought is in the Right Place do you REALLY expect anybody to 
 know what the hell you mean by :

 When you're all locked up, use ^C to
 
 halt and leave the debugger in command, and do the thread apply all bt
 thing. That should be revealing
   
 *Just a thought*
 

 If you want paid-quality tech support...

 pay someone.

 You might want to read this:

   http://www.catb.org/~esr/faqs/smart-questions.html

 if you have just any questions at all about the tone of the
 conversations you see on a technical mailing list on the Internet.

 HTH.  HAND.

 Cheers,
 -- jra
   

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Re: [asterisk-users] What does reason 8 for failure means in Manager

2008-06-04 Thread Al Baker

you mean the CO gave an All-circuts-are-busy tone ???
If not, what does AST_CONGESTION mean

Philipp Kempgen wrote:
 Sanjay Rajdev schrieb:
   
 I tried to call a number on the ZAP channel through manager, I got an 
 Unknown reason for failure, with the following Originate Response. 

 Event: OriginateResponse 
 Privilege: call,all 
 Response: Failure 
 Channel: Zap/G0/ 
 Context: callback 
 Exten: 6563 
 Reason: 8 
 Uniqueid : NULL 
 CallerID : 1234 
 CallerIDNum: 1234 
 CallerIDName: ABCD 

 Can anyone Please let me know what does Reason 8 means here. 
 

 Congestion (AST_CONTROL_CONGESTION).

 Grüße,
 Philipp Kempgen
   

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Re: [asterisk-users] [asterisk-dev] Asterisk 1.6 Realtime Database must use ', ' not '|'

2008-05-24 Thread Al Baker

quote And hackers ignoring pleasantries to get right down to the 
technical issues isn't abusive at all
ABUSIVE - No not at all.
Unnecessarily rude, insensitive, tacky - Yep

Jay R. Ashworth wrote:
 On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote:
   
 This is getting downright abusive, and is totally uncalled for, this
 is not a list for personal attacks.
 

 You thought that Steve suggesting JT step in was abusive?

 If that's not what you meant, then you need to either a) be clearer, or
 b) reply to the proper message.

 And hackers ignoring pleasantries to get right down to the technical
 issues isn't abusive at all. 

 See Jargon File; see also Asperger's Syndrome, How To Ask Good Questions.

 Cheers,
 -- jra
   

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Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.

2008-05-24 Thread Al Baker

Quote 

Oh and also, in my implementation there are no queues. It seems to be
 not related, I've had it in EVERY version of Asterisk I've used.

 Hmmm- maybe this should be mentioned in the next is * Really Good Thread ?


Mark Hamilton wrote:
 Same here.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
 McGowan
 Sent: May 22, 2008 4:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
 havoc.

 Steve Totaro wrote:
   
 On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
 [EMAIL PROTECTED] wrote:
   
 
 Steve Totaro wrote:
 
   
 On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
 [EMAIL PROTECTED] wrote:

   
 
 Mark Hamilton wrote:

 
   
 Hi,

 Yesterday I made a change in queues.conf and so tried doing a reload
 app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
 anything, infact all action on CLI stopped.

 Then, I did a reload. Same thing.

 After that there was no other way.. because even stop now wouldn't
 work, so I did a service asterisk restart

 And then asterisk kept giving the same thing on prompt Died
 successfully and all that it usually says when you issue a stop now,
 except it kept showing that on root prompt after doing a service
 asterisk restart.

 Did a killall asterisk, and finally it stopped.

 Then started asterisk service. It was fine.

 Did a full restart at night, and it was fine.

 NOW, I wanted to do a reload again today mid-day when in full use, and
 it still didn't work, and ALL of the above happened again.

 --

 How do I diagnose what's causing this?

 Thanks,

 Mark.


 
 
   
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 I've had this problem before, haven't debugged it. I definitely look
 forward to hearing what is said about this.

 Example from my recent experience, I wanted to restart the server and
   
 so
   
 did

 pbx0*CLI restart now

 But nothing happened...system continued to allow calls to take place.
 I've found that sometimes exiting and reconnecting to the CLI helps,
   
 but
   
 there have been a couple occasions where NOTHING would allow the server
 to restart save for a reboot. Even killall asterisk didn't kill the
 process

 Sherwood McGowan


 
   
 You are using Asterisk 1.2.x?  I have seen this many, many times.

 Sometimes the CLI becomes unresponsive, sometimes queues crap out or
 stops delivering calls to agents, sometimes it just takes a bit and
 then becomes responsive again.

 The rule of thumb is don't reload queues when there are people in
 queue, at least that seems to eliminate the problems I have seen.
 Makes sense too.

 Not sure if it is fixed in 1.4.

 Thanks,
 Steve Totaro

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 Oh and also, in my implementation there are no queues. It seems to be
 not related, I've had it in EVERY version of Asterisk I've used.

 
   
 I have observed it on repeated general reloads on all versions.
 That's why I don't reload very much, only planned.

 Thanks,
 Steve Totaro

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 My problem exists even when issuing a restart now or stop now command at 
 the CLI.

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-18 Thread Al Baker

Glad I was able to foster some good open discussion.
Hopefully DIGIUM will take to heart some  of the thoughts expressed here
and end up with a BETTER SOLUTION for ALL.

Steve Totaro wrote:
 Inline

 On Fri, May 16, 2008 at 11:44 AM, Tilghman Lesher
 [EMAIL PROTECTED] wrote:
   
 On Friday 16 May 2008 09:11:11 Steve Totaro wrote:
 
 On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher wrote:
   
 On Friday 16 May 2008 06:59:15 Al Baker wrote:
 
 this is one very weak area for *. There is NO ANSWER.
 Now in fairness to *, the answer DOES depend on a # of critical
 variables. How much CODEC to CODEC transcription is going on.
 How many MEET Me conferences are going on.

 On the other hand, DIGIUM COULD, since they have a lab take 4-5
 'standard' workloads
 on two of the most common hardware boxes, say Dell  HP, and run x # of
 transcriptions and
 show the #'s.
 Then x # of meet-me conferences.

 Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks

 Rockwell and NORTEL can tell you this for every piece of hardware they
 sell.

 It is a an area DIGIUM need to man-up in.
   
 I'm not sure what your problem is with Digium.  They sell several
 machines for which they publish very specific numbers as to how many
 users those machines will support (the Switchvox appliances).  Note that
 these machines are configurable only from the web interface, and they do
 not allow you to install additional software.  In other words, when they
 give you a specific machine, with a ton of those variables controlled,
 they can give you a number.

 Digium is under no obligation to give you numbers for your own hardware.
 That's up to you (and you get to control your own set of variables).
 
 It seems any constructive criticism offered, you take as an attack
 against Digium.  That is not a good attitude.
   
 I don't see how you figured out what I was thinking.  Al said Digium doesn't
 publish any numbers, and I responded, saying that he was incorrect; Digium
 does indeed publish numbers (they're just not for his hardware).
 

 I'm not sure what your problem is with Digium.  Proof, period.

   
 While under no obligation, it certainly would help sales.
   
 Whose sales?  If you're talking about the appliances, then yes, I'm sure the
 publication of those numbers help with sales.  If you mean your own sales,
 well, you're right, Digium's numbers probably don't help your sales.  You
 could certainly put together a lab and do your own testing.  Why don't you do
 that?
 

 Sales in general.  You don't need to benchmark everything, just a few
 basic benchmarks, maybe gear it to your hardware and SIP as a gateway,
 then build from there.  Most companies do this.

 I have my own lab and bechmarks but they are for Sangoma hardware and
 very specific servers and all geared to callcenter apps.

   
 I take Appliance Numbers with a grain of salt.  The sales model of
 SwitchVox (and most others) is based on number of ports (SoHO, SMB,
 Enterprise) not maximum number of ports that the appliance could
 actually handle if not artificially constrained.
   
 Consider the maximum number of ports that Switchvox will enable on a single
 machine and consider that the maximum number that they're willing to support
 comfortably without running into some hard limit.  You never want to run into
 a hard limit in the field anyway.
 

 High powered ervers are cheap and so are appliances once you settle on
 an enclosure and guts and start cranking out boxes.  Hard limit
 common.

   
 This is in the style of legacy proprietary systems and anther reason
 why the sale cycle goes a little tougher than a custom job.  Asterisk
 with FreePBX (and maybe Druid) eliminate these artificial constraints
 on usage.
   
 Yes, but the point of those constraints is to permit support a manageable
 job.  Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
 that a particular machine could handle, but from a support perspective, it
 doesn't matter how many the machine could theoretically handle, it matters
 how many it could handle in the particular installation in a supportable
 configuration (those are all those pesky variables we've been talking about).
 

 Maybe that is what the official corporate answer is or, you were
 brainwashed to believe, but I tend to think it is to sell SMB and
 Enterprise software and support.  It is all about money.  I didn't
 fall off the turnip truck yesterday.

   
 I have load averages and CPU usage stats in my mind for all the
 various usages and hardware through experience in my mind.  Of course
 they are only valuable to the exact setup I was doing.
   
 Precisely.

 --
 Tilghman

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Re: [asterisk-users] PBX deployment big problems: Voip traffic analysis

2008-05-16 Thread Al Baker

look - you MUST have a minimum of the following
- a clear 24x7 graph of all you network segments show packet loss, 
packet delay for several weeks prior to 1st turn up.
-Unless you have a 100% totally dedicated IP network for you voice, you 
must have Qos
on every piece of hardware in the network, and you must test it to makes 
sure it works.
As you turn up the service, in controlled stages you MUST measure 
network and systems.
and fix any errors or bad trends in the data.
Or you can say the hell with, just turn it up, and see what happens :)

Bhrugu Mehta wrote:
 hi,
 Yes, there are many problem to implement and setup asterisk in a callcenter.
 but , all these problem can be remove if you set up your hardware and
 your LAN network
 verywell.
 Generaly, your server Configuration should be greater and your LAN also.
 You have to use Proper Codecs for voice. Generaly , g729 is greater.

 regards,
 Bhrugu Mehta


 On 5/16/08, gincantalupo [EMAIL PROTECTED] wrote:
   
 Hi,
  hope not to be OT  :)
  after more than 3 years of PBX installations we can adfirm Asterisk is
  stable enough to be considered a good product but still we encounter a
  lot of problems when deploying a new PBX. It seems that the biggest
  problems are all networking related: one way voice (also inside a LAN),
  calls drops, etc...
  How do you face this kind of problems? Which diagnose tools/methods do
  you use?

  Thank you.

  Giorgio Incantalupo

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-16 Thread Al Baker

this is one very weak area for *. There is NO ANSWER.
Now in fairness to *, the answer DOES depend on a # of critical variables.
How much CODEC to CODEC transcription is going on.
How many MEET Me conferences are going on.

On the other hand, DIGIUM COULD, since they have a lab take 4-5 
'standard' workloads
on two of the most common hardware boxes, say Dell  HP, and run x # of 
transcriptions and
show the #'s.
Then x # of meet-me conferences.

Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks

Rockwell and NORTEL can tell you this for every piece of hardware they sell.

It is a an area DIGIUM need to man-up in.

Alexey Shimeshov wrote:
 Hello, Alexander.

 AO Hi Asterisk Users,

 AO I'm interested in how many concurrent calls Asterisk can process without
 AO troubles. I mean 1 Asterisk server (software) like either proxy or media
 AO server (any numbers will be appropriate).

 AO 1. Is there any limitations by the software? What is this number?
 AO 2. What is the maximum count of concurrent calls you've ever seen/tested?

 Look at this example

 http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm

   

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Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI

2008-05-16 Thread Al Baker

This is 'basically' a tie-line between the boxes.
Yes - it is done all the time between PBX's. You are basically nailing 
up a circut between the boxes.
It could be a simple as a simple POTS leased line or a multi-t1 bundle 
between them.
How it is physically done with DIGIUM's boards under * ?

Someone else will have to answer that


Pascal Maugeri wrote:
 Hi

 I have a system (S) that has a PSTN gateway to accept incoming calls 
 and setup outgoing calls from/to Telco network. In the other hand I 
 have a distant Asterisk box (A) that I would like to connect to (S) 
 using the PRI interface.

 I understand that the proper way is to order to my Telco two PRI lines 
 one for (S) and another for (A), and configure (S) and (A) to call 
 each other numbers when they have to interconnect.

 Now, might it be possible to connect directly (A) and (S) using their 
 PSTN interfaces without having to go through to my Telco ?! Does it 
 make sense ? Is it technically feasible ? I guess that the Telco 
 network is providing routing, number assignation, etc. and it sounds 
 pointless to do this. Nevertheless could you confirm it is 
 possible/impossible and why ? Is there a better way to do that ?

 Thanks in advance,
 Pascal


 

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Re: [asterisk-users] Asterisk for Larg

2008-05-15 Thread Al Baker


Whoa - you need some highly reliable, TELCO quality iron with some 1st 
class support for that.

Do you realize what your downtime in that environment would would cost 
you ?

Look, * is cool , fun an customizeable etc.

But it IS NOT carrier grade hardware and it is NOT software produced in 
Certified Software Enviroment with a Certified CMM rating.

Yes -its cool and neat and cheap.

But for that big of a set up you need a whole different kind of solution



 Is Asterisk practically stable and reliable for a larg Enterprise has say a
 1 phones , is there any case study like this?
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Re: [asterisk-users] New Asterisk Deployment - Need some tips

2008-05-15 Thread Al Baker

The items most people do not address are:
- QA - How do You tell if you you having Jitter,Packet Loss etc BEFORE 
the user scream
- Disaster Recovery - from the small - DNS smokes - To Larger - * box 
with 96 ports smokes
- Insuring EACH and EVERY piece ox network SUPPORT and USES QoS
-Vendor SLA - How do YOU measure the service, WHAT happens outside 9-5
-HW Support - Your Quad port DIGIUM card smokes. Can you live w/out it ? 
Should you have a spare on hand ?
If so how many
-What TOOLS are you going to use to MONITOR this whole thing - all 
servers, switches
-800 Phones - Minimum . Could be painful if folks are used to 
traditional TELCO reliability and Quality

Andrew Latham wrote:
 Ditto.

 If you need to quantify the consultant to the powers that be just ask
 for an Infrastructure Audit.  I have done several in the past that
 have saved tons of money that encouraged further phone projects.
 Finding dead phone lines to discovering unused but rented telcom gear
 is always fun.  Also when setting up you test group make sure they
 actually use the phone and often...



 On Wed, May 14, 2008 at 9:32 AM, John Signorello [EMAIL PROTECTED] wrote:
   
  I would have to agree with Grey Man, a pilot project is one way to start
 up.

  I would also seriously recommend buying some consulting time from an
  experienced Asterisk PBX vendor/dealer/consultant.

  The cost is negligible in light of the scope of your project.

  A pilot project will only give you a glimpse of what is required.

  You have to have a design that incorporates your eventual build out.
  A pilot by itself is not going to give you that. You will need help from
  a source that can bring their experience to help you tip toe around the
  potential land mines you can encounter.

  regards,

  John Signorello
  Managing Partner
  ispbx.com
  866 GO ISPBX



  Grey Man wrote:
  On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff
 [EMAIL PROTECTED] wrote:


  I'll be doing a new Asterisk deployment soon, and would like to gather your
 thoughts.

 Here are some items that need to be kept in mind:

 Support 800 phones (400 of which are analog)
 Concurrent calls ... ? but need to guess high so that the server can handle
 this.
 Voicemail will be required along with sending voice mail attachments to
 email server.
 Flash panel for switchboard operator.
 Needs to be a distributed server design for redundancy and fail-over.
 Will need to be integrated into an existing PBX until each building is
 switched over to use the Asterisk servers.
 If calling 911 from a building among multiple buildings, how can EMS find
 that person based upon the call?
 What type of data line should be used in this setup? T1?
 The physical network will support QOS and the like, so that is not an issue.


 What type of design/setup do you recommend for this? How about server
 resources...ie...CPU, RAM, Disk space.

 How about backups? Does imaging work best if a server were to fail?

 Any thing else you can think of?


  If this is a project for your work and it's your first Asterisk
 deployment then definitely don't go the big bang approach in the way
 you've outlined. If you do you could well be out of that job in 6
 months!

 The first thing I'd recommend you do is find 10 or 20 people who are
 suitable as early adopters. The set up a single Asterisk server and
 give the early adopters a SIP phone each thats in addition to their
 normal desk phone and ask them to see how they go using the SIP phones
 for calls to each other, external calls and whatever else would make
 sense. Then 6 months and a lot of learning/experience/frustration
 later you'll know whether to get answers to your original questions or
 not.

 Regards,

 Greyman.

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Re: [asterisk-users] QOS and Asterisk

2008-05-15 Thread Al Baker

You SHOULD be concerned with QOS. All the way to an including the vendor 
or your service cold really sucku

Michael Graves wrote:
 On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote:

   
 I will have a small shop with ~4 phones using an HP server with Asterisk on 
 it, it has two NICS and so I planned on plugging one into the cable modem, 
 and the other into the switch. I was going to let this box perform NAT for 
 the company but I am concerned about QOS for the VOIP portion.

 Anyone got a similar setup and care to share what they successfully 
 implemented?

 Thanks!
 jlc
 

 You should take a serious look at Astlinux. It's en embedded Asterisk
 distro that handles routing, including QoS, when necessary. See
 www.astlinux.org.

 Michael
 --
 Michael Graves
 mgravesatmstvp.com
 http://blog.mgraves.org
 o713-861-4005
 c713-201-1262
 sip:[EMAIL PROTECTED]
 skype mjgraves
 [EMAIL PROTECTED]



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Re: [asterisk-users] Installation Question

2008-05-13 Thread Al Baker

I thought that the point that you had to have a timing source for *.
That source could be the clock off the T-1.
But if you didn't have something like your T1 to provide master clocking
ztdummy was something to provide the required a source for timing.

Joseph L. Casale wrote:
 Sure if you don't need ztdummy, or is there a newfangled way around that?

 Thanks,
 Steve Totaro
 

 Hi Steve,
 I read the wiki and see this provides timing for Asterisk. Can you point
 me toward a description of what exactly this does? I was checking out the
 tutorial at 
 http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm
 and noticed they never compiled either this or the Libpri which is what 
 prompted
 me to assume I may not need it in my scenario.

 Appreciate the help!
 jlc

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Re: [asterisk-users] 3U server chassis Digium TE405P?

2008-05-12 Thread Al Baker

Getting the RIGHT card for the RIGHT bus type and the RIGHT Chassis is 
NOT as simple
as everyone will lead you to believe.

My suggestion, worth exactly what you paid for it :)

Get Exact Spec for the card your are considering and FAX / Email to PC 
vendor and have him
send you In Writing that the card WILL fit in the box and in the bus.

Then I would get the Exact Spec for the BOX and BUS in the box and send 
to DIGIUM or their OEM
and get THEM to tell you it should all work.

Overkill - some will say yes.
But
THEY won't be sitting there with you if your expensive Server comes in 
and your expensive   card come in an they no-workie together

Sherwood McGowan wrote:
 Matt Watson wrote:
   
 I'm not sure if a full-height card would fit (vertically) in a 3U chassis... 
 but I would probably also assume that if it would not, that the chassis/mobo 
 would have a PCI/PCI-Express riser card that would mount the cards 
 horizontally.

 Might want to check that out with the manufacturer of the chassis.

 --
 Matt

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood 
 McGowan
 Sent: Monday, May 12, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] 3U server chassis  Digium TE405P?

 Gentlemen,

 First let me say it's great to be back on the Asterisk mailing lists.
 Those of you who have been around for a while will remember me as
 Rushowr. I look forward to answering questions and whatnot in the
 future, but for the moment I have a minor question that I cannot find a
 definitive answer for online.

 I am in possession of a Digium TE405P card which I _know_ will fit in a
 4U chassis, but we are building a new server and cannot get a 4U from
 the supplier that my current client wants to use. However, we can get a
 3U chassis. My question is, will this card fit? Does anyone out there
 have a 405 out there that they have installed in a 3U?

 Thanks in advance for any help that can be offered,
 Sherwood McGowan
 VoIP / Telecom Solutions Consultant

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 Thanks for the heads up, I've found full height capable 3U chassis. The 
 worst thing about this whole ordeal was that I assumed (very bad idea, 
 of course, stupid stupid stupid) that the 2u server had a riser card, 
 which it did not :( Ah well, live and learn...

 Sherwood McGowan

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Re: [asterisk-users] Which sound file formats?

2008-05-12 Thread Al Baker

Asterisk will automatically chose the best format  - per ATFOT

Roderick A. Anderson wrote:
 I've got the text files created -- thanks to Russell Bryant -- for 
 re-building the core and extra sounds using another voice but I'm not 
 sure which formats to actually build.

 This will be a small/personal system using Vitelity.net so will only 
 have SIP connections.

 The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, 
 .gsm, .ulaw, and .wav.

 What are the minimal formats I need or can get by with.  Possibly even 
 an ordered preference list.


 Thanks,
 Rod
   

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Re: [asterisk-users] func_odbc creating records or best practice


Quote 

func_odbc can do whatever queries you give it.  SELECT/UPDATE are
simply the simplest cases that make it easy to understand the functionality

*OK - but are the Limited to SINGLE STATEMETS or can you have a Muli-Statemnt 
Transaction ?*?

Tilghman Lesher wrote:
 On Monday 28 April 2008 17:30, Robert McNaught wrote:
   
 I am trying to write a custom application which will integrate with an
 existing MSSQL crm system.

 We need to get ahold of the CDR(uniqueid) field in during call-time -
 I see from doing a DumpChan(), the CDR unique ID is available as soon
 as the call is created.  CDRs usind odbc are only written once the
 call is completed.  Does anyone know if it is possible to use
 func_odbc to create a temporary record then delete it so that this
 information is available to MSSQL.  I was not sure if func_odbc was
 limited to just using UPDATE/SELECT queries.
 

 func_odbc can do whatever queries you give it.  SELECT/UPDATE are
 simply the simplest cases that make it easy to understand the functionality.

   
 Would there be a better way to do this using the AMI or AGI?  It just
 seems a little strange to use a database for storing temporary data
 such as this?
 

 I'd agree with you on that.  I would tend to set variables directly in the
 channel, then query them out using AMI.

   

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Re: [asterisk-users] func_odbc creating records or best practice

I would love to be able to issues the necessary Mysql commands to have 
true TRANSACTIONS
Such as - Begin Transaction
   Select @var=agent.id, agent.exstension where 
agent.status='free'
Update agent.status='BUSY' where [EMAIL PROTECTED]
End Transaction
Of Course the syntax I used above is just psuedo-code and NOT correct MySQL
but I think you can see what I am trying to do. Which I think would be 
darn handy !!!

Tilghman Lesher wrote:
 On Friday 09 May 2008 01:39:53 Al Baker wrote:
   
 Quote 

 func_odbc can do whatever queries you give it.  SELECT/UPDATE are
 simply the simplest cases that make it easy to understand the
 functionality

 *OK - but are the Limited to SINGLE STATEMETS or can you have a
 Muli-Statemnt Transaction ?*?
 

 As we don't isolate connections to a single channel, we do not support
 multi-statement transactions, no.  It's an interesting idea, though.  Could
 you expound on what you would like to see?  It may wind its way into a
 future version of func_odbc.

 Perhaps only three extra statements, one to start a transaction, which
 also reserves the connection handle to the channel (note that this would
 require turning off connection sharing, which is the default, except for TDS
 databases), one to commit, and one to rollback (the last two here would
 also release the connection handle back to the pool).  Would that be
 sufficient?

   

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Re: [asterisk-users] Asterisk in Production ?

Thank you for your very kind offer.
After repeatedly re-opening the ticket I finally got a clear specific 
answer.
Strangely, in the 30 mins it took for me to take the answer, try it, and 
report back the results
they had closed the ticket again so I couldn't report whether their 
solution fixed the problem or not.
In fact it did, but I would have liked to have been able to document 
that so that others running into
the same problem and scanning the bug report would know definitively if 
their answer was indeed correct.

But - THANK YOU - and I will Certainly take you up on your most kind 
offer in the future!

Tilghman Lesher wrote:
 On Thursday 08 May 2008 23:38:14 Al Baker wrote:
   
 Take a big shot of Valium before dealing with the bug tracker folks.
 There idea of help is to post You have an extra space in your line
 then CLOSE the ticket.
 That kind of clear, specific help is just what my doctor ordered to keep
 my BP nice and low
 

 If you have a problem with one of the explanations, please post the bug
 number here, and I'll be happy to explain it in more detail.

   

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Re: [asterisk-users] Best Linux distribution to use in Asterisk server

this often becomes a religious discussion.
my free advice worth all you paid for it - Redhat or one of the other 
distros that has been Certified on your choice of hardware and which has 
a Support Contract on  it. Despite what others will tell you... Its a 
lonely place when your box no-workie and you have no support contract.
if you buy Redhat on an HP box that is certified for it. It WILL work or 
they WILL get it working.
sure maybe someone somewhere on some mailing list has the answer. But 
you got 96 lines down  with a box
and customers screaming... You want to hope, that maybe, someone 
will respond an respond correctly to
your problem on a mailing list, or call a Customer Support Center 
staffed 24x7 with engineers trained on your specific hardware and you 
specific O/S ?




Philipp Kempgen wrote:
 equis software schrieb:
   
 Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
 you think about to use Ubuntu or another distibution??
 

 I prefer Debian, but if everything works well and if you're
 familiar with Gentoo why change?


 Grüße,
 Philipp Kempgen
   

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Re: [asterisk-users] Zap Channels Collide (Incoming Outgoing)

I know that everyone has gaps in their knowledge, but I am just 
staggered that
systems are being sold/deployed with such fundamental TELCO workings not 
being
understood.  Frightening.

C. Chad Wallace wrote:
 At 5:22 PM on 08 May 2008, Forrest Beck wrote:

   
 I have a client that is using the Sangoma A200DE with two phone
 lines attached.

 The problem is:

 They use their phone (Grandstream GXP2020) to dial out of the system.
 Instead of getting ringing, there is someone on the other end of the  
 line that happened to dial in at the exact same moment.

 So now they are stuck talking with this person, instead of the one
 the originally called.

 The ZAP channels are in a dial plan context that instructs it to
 just dial the office phones.

 [zap1]
 exten = s,1,Dial(SIP/1001SIP/1002SIP/1003)
 exten = s,n,Voicemail([EMAIL PROTECTED])

 Anyone know how to get around this?
 

 This is known in the telephony world as glare, and there's not much
 you can do about it, especially if you only have one line.

 If you have multiple lines on an over-ring (or hunt group or whatever
 you call it), the best thing to do is find out which way the telco
 assigns calls to those lines wrt how they are assigned to the Asterisk
 box.  And then allocate outgoing calls in the other direction.  

 On our installation, the calls are allocated from the first FXO port
 (Zap/25) up.  So we set Asterisk to dial out starting from the last FXO
 port in the group by calling Dial(Zap/G2) (capital G means dial down
 from last, lowercase g means dial up from first).  That minimizes glare.

 But, as I said before, if you only have one line, you can't do that...

   

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Re: [asterisk-users] AGI asterisk high balance

I think his connect/disconnect is going to take far longer than his 3 
queries.
The fact that Asterisk doesn't support sustained MySQL connection from 
the DialPlan
is in fact quite a big deal that Digium seems to have its head in the 
sand about.
And one of those things that SHOULD come up in those Is * Ready For 
Prime Time Threads

Rizwan Hisham wrote:
 Well database really is a bottleneck for me. I am currently trying to 
 do rating stuff in agi using perl. What im doing is i lookup the rate 
 of every dialed code for every call from the mysql database using the 
 longest match technique. The longest match technique costs atleast 2-3 
 mysql queries for every call untill the dialed code is matched out of 
 14000 dialcodes. I dont know how to calculate the exact delay due to 
 execution of agi, but on the asterisk cli whenever that agi executes, 
 there is a visual delay of about half a sec to move from the agi 
 extension to the next extension (can anybody tell me how to calculate 
 the delay).

 Now im planning to use the manager api for constant connectivity to 
 mysql and to enhance the longest match technique. Can anybody help me 
 with this? Is it a good idea to ue manager api for  lookingup the rate 
 of the live call?

 On Sun, May 4, 2008 at 1:34 PM, Grey Man [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 If you've got anything but trivial AGI loads you should switch to
 FastAGI and put your business logic on a separate server to your
 Asterisk server. I use a deployment where a call could make up to 3
 AGI requests per call before being put through (for things such as
 looking up accountcode, checking account credit, setting PSTN
 callerid). We monitor the time thw whole process takes and on average
 it's less than 100ms on an Asteisk server that peaks at 200
 simultaneous calls (400 bridged) and 3 to 5 call set ups per second.
 The business logic processing the FastAGI   calls is C# and .net which
 means Java would be able to handle it easily as well. The most likely
 bottleneck under high load will be your database.

 Regards,

 Greyman.

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 -- 
 Best Regards
 Rizwan Hisham
 

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Re: [asterisk-users] Asterisk in Production ?

Perhaps this should be tagged under Is * Ready For Prime Time ? Thread
Isn't an 'appliance' supposed to be a 'plug-it-in-and-runs' sort of thing ?


Julian Yap wrote:
 On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
   
  We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
  and it's quite unstable.
  We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy
  deadlock
  and now that we have added a Queue, it's worse than ever. The queue goes
  stuck quite often
  (agent are stuck in 'In use' state and if they logoff they can't log-in
  till an asterisk restart).
 

 There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1.

 There's another thread on this.

 - Julian

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Re: [asterisk-users] Asterisk in Production ?

Take a big shot of Valium before dealing with the bug tracker folks.
There idea of help is to post You have an extra space in your line
then CLOSE the ticket.
That kind of clear, specific help is just what my doctor ordered to keep 
my BP nice and low

Benoit Plessis wrote:
 Tilghman Lesher a écrit :
   
 On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
   
 
 lordfuknowsyou a écrit :
 
   
 Vinícius Fontes wrote:

 I use 1.4.18 with no problems. We have quite a few users(125 total
 between branches), but the call volume at the most has been around 15
 active calls at a time.
   
 
 Any IAX2 phone or mostly SIP ?
 Do you use Call Queues ?

 We have less user than that, less concurrent call but quite a few
 crash/deadlocks
 
   
 Have you reported these issues on the bugtracker?

   
 
 Well, the problem is finding usefull data to report.

 I've 4 core dumps thats show differents things:

 two seems to be related to ControlPlayback:
 #0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
 #1  0x0809c579 in ast_readframe ()
 #2  0x0809defc in ast_streamfile ()
 #3  0x0805e786 in ast_control_streamfile ()
 #4  0xb698be5c in ?? () from 
 /usr/lib/asterisk/modules/app_controlplayback.so
 #5  0x08298700 in ?? ()
 #6  0xb470aec0 in ?? ()
 #7  0xb698c1fc in ?? () from 
 /usr/lib/asterisk/modules/app_controlplayback.so
 #8  0xb698c1fa in ?? () from 
 /usr/lib/asterisk/modules/app_controlplayback.so
 #9  0x in ?? ()
 

 One is pretty generic:
 #0  0x0809c9bc in ast_closestream ()
 #1  0x08085d91 in ast_hangup ()
 #2  0x080cd3d8 in pbx_builtin_setvar_helper ()
 #3  0x080cf08e in ast_pbx_outgoing_exten ()
 #4  0x080fde65 in ast_inet_ntoa ()
 #5  0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0
 #6  0xb703667e in clone () from /lib/tls/libc.so.6


 and the latest is thread/iax2 related:
 #0  0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
 #1  0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
 #2  0x0079 in ?? ()
 #3  0x in ?? ()
 #4  0xb547a148 in ?? ()
 #5  0x080f0508 in ast_sched_add_variable ()
 #6  0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
 #7  0x0012 in ?? ()
 


 But my main problem is when the system just froze,
 it start mostly by the Queue not working anymore, with member stuck in 
 'in use' stack (should not happen
 with IAX2 agent IIRC, given that we had to build macros using GROUP() to 
 detect in use IAX2 agent)
 Then the console (asterisk -rcTvvv) start to freeze (completion doesn't 
 work, message stop from being displayed
 and even command output is lost).

 And i'm reading http://www.asterisk.org/developers/bug-guidelines which 
 speak of using SVN trunk version of asterisk,
 thing i'm not really eager to try on a live system...




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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

2008-05-06 Thread Al Baker

Are you saying the * server does NOT TRY to re-establish the BD connection ?

Does your whole * SERVER freeze ?

If  NOT, what happens to you CDR records ?

Anthony Francis wrote:
 Tilghman Lesher wrote:
   
 On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
   
 
 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
 
   
 On Monday 05 May 2008 11:24, Johansson Olle E wrote:
   
 
 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
 
   
 On Monday 05 May 2008 09:45, Johansson Olle E wrote:
   
 
 Another issue that we need to fix with the MYSQL driver is that
 we're
 lacking a connection pool. Everything seems to be handled over one
 connection to Mysql, which causes issues.
 
   
 That's not true.  The MYSQL app generally uses multiple connections,
 one
 for each channel.  The only way one might use only a single
 connection is
 by using a global variable to store a single connection id, but that
 method
 is not documented anywhere, AFAIK.
   
 
 You talk about the Mysql APP, but is this the case with the Realtime
 driver as well?
 
   
 No, the native Realtime driver uses a single connection.  The ODBC
 Realtime
 driver generally uses a single connection but can be configured to
 use a
 separate connection for each query.
   
 
 So, we're back to where we started. A developer that can help us with
 a connection
 pool or a separate connection for each query would be a Nice Thing (TM).
 
   
 What issues are you specifically seeing that merit using multiple
 connections?

   
 
 I can specify an issue that would merit multiple connections, if the 
 link to your db goes away Asterisk likes to freeze writing CDRs.
 I have a few remote * servers that this happens to. My solution so far 
 has been to record CDR's to a local DB and then have a
 perl script that attempts to move them over to my transaction DB. I 
 would suggest this solution to anyone who depends on their CDR records.

   

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

2008-05-05 Thread Al Baker

I looked all over VOIP-INFO and ATFOT and could not find anything that 
said or even suggested not using the mysql driver.(except NOT to have 
BOTH drivers loaded at the same time). I could easily be missing 
something. But the apparent BUG I am seeing is at such a Basic and 
Simple Level of functionality that either DIGIUM ought to fix it ASAP or 
update VOIP-INFO pages and their own documentation to say Broke - No 
Workie and We Are No Gonna Fixie :)

Steve Totaro wrote:
 On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
   
  I would appreciate any and all advice on what appears to be a BUG (or a
 brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
 fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1

  It appears that the interface eats the first field requested from a
 table. If only One Field is Requested from the Table , that field is eaten
 ENTIRELY by Asterisk. If several fields are requested, the First Field Is
 Eaten and the remaining filed are returned, but place in the WRONG Variable
 since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
 Different ways to approach this.

  Extension  – Get only ONE (1) field from Table

  Extension  – Get THREE(3) fields from the Table and Quote Them.

  Extension  - Get THREE(3) fields from the Table

  I have show the Output from the Asterisk CL for each, which clearly show
 that SOMETHING is not
  right. Maybe the Software, maybe the person using the software :)

  Here is the Table in the Database.

  mysql select * from agent;

  +--+-+++-+

  | id  | cust_id  | status |phone |tlce |

  +--+-+++-+
  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |

  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |

  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |

  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
  +--+-+++-+

  4 rows in set (0.00 sec)


  Here is the DIALPLAN

  exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)

  exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
 status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')

  exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)

  exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
 ${mytlce})

  exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
 CONNID is ${connid})

  exten = ,n,MYSQL(Clear ${resultid})

  exten = ,n,MYSQL(Disconnect ${connid})

  exten = ,n,HANGUP



  exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
 'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')

  exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)

  exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
 ${mytlce})

  exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
 CONNID is ${connid})

  exten = ,n,MYSQL(Clear ${resultid})

  exten = ,n,MYSQL(Disconnect ${connid})

  exten = ,n,HANGUP


  exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)

  exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
 agent\ where\ phone=\'1234567890\')

  exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)

  exten = ,n,NoOp(CUSTID is ${custid})

  exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
 CONNID is ${connid})

  exten = ,n,MYSQL(Clear ${resultid})

  exten = ,n,MYSQL(Disconnect ${connid})

  exten = ,n,HANGUP




  Here is the Asterisk CLI Output

  dial 

  == Console is full duplex

  *CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
 localhost ivr ivrxxx dtc) in new stack

  -- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
 cust_id from agent where phone='1234567890') in new stack

  -- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6 
 custid) in
 new stack

  -- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is ) in new stack

  -- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT ID 
 is
 .. 6 CONNID is 5) in new stack

  -- Executing [EMAIL PROTECTED]:6] MYSQL(OSS/dsp, Clear 6) in new stack

  -- Executing [EMAIL PROTECTED]:7] MYSQL(OSS/dsp, Disconnect 5) in new 
 stack

  -- Executing [EMAIL PROTECTED]:8] Hangup(OSS/dsp, ) in new stack

  == Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'

   Hangup on console

  *CLI dial 

  == Console is full duplex

  *CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
 localhost ivr ivrxxx dtc) in new stack

  -- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
 cust_id, status, tlce from agent where phone='1234567890') in new stack

  -- Executing [EMAIL

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

2008-05-05 Thread Al Baker

I must be overlooking it, I pulled up the electronic version and 
searched for and read every instance where ODBC was mentioned and I 
could not find a single place where it said
ODBC was to be the only or even the best method.  If so I would never 
ever have gone down this road :(

Quote And according to the O'Reilly book ODBC is the way to go.

Roderick A. Anderson wrote:
 Steve Totaro wrote:
   
 A quote from Tilghman Lesher from a previous post.

 That's fine, but I have had the most horrid results using any distribution-
 supplied ODBC drivers.  The best results are obtained by source-compiling
 the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
 PsqlODBC.  UnixODBC is fairly safe to use from distribution channels, 
 however.
 

 And according to the O'Reilly book ODBC is the way to go.

 Though they use PostgreSQL for their examples and Asterisk is installed 
 on a CentOS system the instructions are really good.  Getting it to work 
 with MySQL should be pretty simple and I'm sure on-line resources for 
 doing this are be out there.

 soapbox
 Personally I never use MySQL except in cases where I am under extreme 
 duress.  Therefore I tried and tossed trixbox, AsteriskNOW, and 
 freeePBX.  Yes I know I can get around the database engine issue but 
 that is what a distribution should be for: no hacking (or at least 
 not-too-much) required.

 It is now CentOS 5, Asterisk from source, PostgreSQL (on another system) 
 and hand edited (for now anyway) *.conf files.  Maybe AsteriskGUI later.
 /soapbox


 Rod
   

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Re: [asterisk-users] Minimum upload speed for Asterisk?

2008-05-01 Thread Al Baker

You also need to check for Packet Loss on the Link

Erik Anderson wrote:
 On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote:
   
  Is 384kB up too slow?
 

 Probably not.

   
  Is there any guidance for the minimum upload speed for an Asterisk box?
 

 I'm guessing this is for just a few calls at a time, correct? I'd
 guess that rather than these quality issues being caused by cramped
 bandwidth, they're actually being caused by latency issues.  Have you
 ever checked the latency of the connection between your asterisk
 server and your SIP/IAX endpoint? If it's really high (say 300ms+) or
 if the latency is really erratic, you'll have quality issues.

 You didn't mention whether you are doing traffic shaping on your
 upstream connection, so I'll assume you're not.  That would be
 something good to look into - with traffic shaping, you can prioritize
 your VoIP traffic over all other types of network traffic.

 -erik

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Re: [asterisk-users] need examples of asterisk and mysql integration

Why would you go to the trouble of writing a PERL AGI and take the 
Performance Hit of using AGI as opposed to using the  built-in  MYSQL 
from the dial plan ?

Mike Trest - On Travel wrote:
 Hi,

 I suggest you look at writing a PERL  agi program to handle all of 
 the MYSQL / DB
 access and just pass variables between your CONTEXT/dialplan.   I have done
 a lot of these things.  You can get PERL examples for DBI  and use one of
 provided  agi scripts as a prototype.

 ..mike..

 At 04:13 PM 4/22/2008, you wrote:
   
 I'm presently working on a project to build a scheduling system
 accessible by both web and phone.  on the web side one can query what
 items are available when by using the time or the item as a key then
 reserve for an available time slot.  reservations may also be modified
 by the user that made them or an admin.  Where may I find examples of
 doing similar things with asterisk?  all I've been able to find thus
 far is examples of how to store call detail records and voicemail
 using a database.

 Thanks in advance,

 Eric

 P.S.

 Has anyone already built an asterisk/web based scheduling system like this?

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Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue

Why would you want a channel to continue after the caller has hung up.
I clearly am missing something here because I can't see what good that
would be.  What do people do with this Continued Channel ?
What is is used for ? How Does having it help you ? ???

Atis Lezdins wrote:
 Queue will continue if called person hangs up (and there's no option).
 If caller hangs up, call goes to h extension in same context. Just the
 same way as Dial with 'g'. There's a change in 1.6 that allows called
 channel to continue if caller hangs up, so probably something like
 this could be applied also to Queue (or was that actually working with
 using Local channels?).

 Regards,
 Atis

 On Wed, Apr 23, 2008 at 7:13 PM, AnDY [EMAIL PROTECTED] wrote:
   
 Thank you for your answer.
  But the Dial command has a option 'g' which means that after succes will
  proceed next priorities in the dialplan. Is there something also for
  Queue() because according to manual there is no option for it. So I am
  looking for some other solution.

  Andy

  Tony Mountifield napsal(a):


 
 In article [EMAIL PROTECTED],
   
[EMAIL PROTECTED] wrote:
  
   Hello everybody.
  
   I was looking for the solution but nothing found. I have this in my
   extensions.conf:
  
   exten = 233,1,SetAccount(queue1)
   exten = 233,2,Queue(queue1|rn)
   exten = 233,3,NoOp(${QUEUESTATUS})
   exten = 233,4,NoOp(${DIALSTATUS})
  
  
   But when the call is placed in the queue and somebody answer it, it will
   throw an error:
 == Spawn extension (default, 211, 4) exited non-zero on
   'Local/[EMAIL PROTECTED],2'
  
   And no other command in extensions is executed.
   Any suggestions?
  
  
   Queue() is like Dial(), in that if it succeeds in connecting to someone,
   it will not return to the next priority in the dialplan. However, if you
   define an 'h' extension, that will get executed when the call is complete:
  
   exten = 233,1,SetAccount(queue1)
   exten = 233,2,Queue(queue1|rn)
   exten = 233,3,NoOp(${QUEUESTATUS})
   exten = 233,4,NoOp(${DIALSTATUS})
  
   exten = h,1,NoOp(${QUEUESTATUS})
   exten = h,2,NoOp(${DIALSTATUS})
  
   Cheers
   Tony
  




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Re: [asterisk-users] need examples of asterisk and mysql integration

I did not mean to stir up a hornet's nest or religious war :)
The BASIC QUESTION I was trying to ask is this...
   Since the MYSQL add-on provides a way to interface with MySQL
 what is it that one gains or is trying to gain by writing their OWN
 AGI script to do the interface ?

The only reason I mention this being a performance hit was that the 
book ATFOT
mentioned that AGI scripts in general were less efficient and therefore 
one should
do all possible work in the dial plan.

Maybe this is true, maybe this is not true.

Regardless.

I would just like to borrow from the expertise and experience of other 
and ask the BASIC QUESTION above and hope that folks will share their 
experiences. I'm pretty sure they did what they did for very very good 
reasons that I am just not experienced enough yet to know.

So The answer to the questions is ???



Steve Edwards wrote:
 On Wed, Apr 23, 2008 at 11:07:01AM -0700, Steve Edwards wrote:
 

   
 AGIs do not have a substantial performance hit and I think people need
 to get this misconception out of their heads.

 Writing AGIs in a scripting, non-compiled language may be great for
 prototyping and proving concepts where performance is not expected to be
 an issue. Personally, I don't write AGIs in anything but C. It's the
 sharpest tool in my kit and I know it best.
   

 On Wed, 23 Apr 2008, Tzafrir Cohen wrote:

   
 The dialplan isn't compiled either.
 

 I guess I didn't make that point forcefully enough.

   
 Now, if it is simple enough to be implemented with the dialplan, then 
 you don't need to execute half the logic in a separate environment and 
 don't need the performance hit.
 

 I'd like to put this performance hit thing to rest. I respect your 
 knowledge and expertise -- I always seem to learn something new (to me) 
 from your posts.

 How would you quantify the hit? On my recently retired dev box (1.6GHz 
 Celeron), I could execute 100 AGIs per second but that doesn't really 
 answer the question.

   
 Anyway, what about dialplan logic in lua? Anybody actually uses that 
 thing?
 

 Oh great -- another scripting language to learn :)

 Thanks in advance,
 
 Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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Re: [asterisk-users] need examples of asterisk and mysql integration

Thx so much for taking the time to share.
Damn Insightful   Damn Helpful
THANKS!

Steve Edwards wrote:
 On Wed, 23 Apr 2008, Al Baker wrote:

   
 The BASIC QUESTION I was trying to ask is this...
   Since the MYSQL add-on provides a way to interface with MySQL
 what is it that one gains or is trying to gain by writing their OWN
 AGI script to do the interface ?
 

 I like doing serious work in an AGI instead of the dialplan because:

 1) It allows me to use a compiled language. Not just for performance 
 reasons, but because a compiler (or a strict interpreter) helps me protect 
 me from me. My production dialplan is just over 600 lines. The sources to 
 my AGIs are a bit more than 15,000 lines. I like that if I fat-finger a 
 variable, the compiler will help me. If I fat-finger something in 16,000 
 lines of dialplan will I ever find it?

 2) It allows me to hide complexity. I like having a single statement in my 
 dialplan that says agi(block-ani). I know that in this single statement 
 I am invoking code that acts as the gatekeeper to my system, allowing me 
 to block callers by area code, area code and prefix, and the complete 
 subscriber number. I know it works well and I don't have to look at it any 
 more.

 3) I can share better. It is easier to integrate a single statement into 
 an existing dialplan than 2,000 statements with potential conflicts in 
 context, template, and variable names.

 4) I don't have to learn (what appears to me to be) a really obtuse 
 syntax. Funky quoting and whitespace rules lead to accelerated hair loss.

 5) I have a full toolbox. For instance, one of my AGIs (play-path) lets 
 me pass a path and it returns the name of a WAV file in that path at 
 random. Calling ftw() was a simple solution. Another AGI (auth-card) lets 
 me submit an authorization request to my credit card processor. While 
 waiting for the card response, I play Please wait while we validate your 
 card in another thread in the same AGI. By the time the STREAM FILE is 
 finished, I have the response so to the customer it appears 
 instantaneous.  How would I do either of these in dialplan?

 Thanks in advance,
 
 Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?

Please keep us updated on your progress.
I am considering putting several of these boxes in
and I would love to hear how this comes out.
Wish I had something to suggest.

Ex Vito wrote:
   Hi list,

   After a lot of testing + troubleshooting, I guess I'm observing
   what I am now calling a regression with zaptel 1.4.10 (is it?)
   As such I call for peer feedback, before either asking Digium
   install support or filing a bug.

   Thanks in advance!


   System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card
   OS: Centos 5
   Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5)
   HW: Digium TE220B, the one with HW echo cancellation
  (configured as 2x E1 via jumpers)

   Context: Pre-site installation of system, no E1 conectivity
(loopbacks tested)


   /etc/zaptel.conf:
   span=1,1,0,ccs,hdb3,crc4
   bchan=25-39,41-55
   dchan=40
   span=2,2,0,ccs,hdb3,crc4
   bchan=56-70,72-86
   dchan=71


   Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel
   buffer:

 About to enter spanconfig!
 Done with spanconfig!
 About to enter spanconfig!
 Done with spanconfig!
 About to enter startup!
 TE2XXP: Span 1 configured for CCS/HDB3/CRC4
 timing source auto card 0!
 wct2xxp: Setting yellow alarm on span 1
 timing source auto card 0!
 SPAN 1: Primary Sync Source
 VPM400: Not Present
 VPM450: echo cancellation for 64 channels
 BUG: soft lockup detected on CPU#0!
  [c044d448] softlockup_tick+0x96/0xa4
  [c042ddc8] update_process_times+0x39/0x5c
  [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c
  [c04059bf] apic_timer_interrupt+0x1f/0x24
  [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp]
  [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp]
  [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp]
  [c042621c] release_console_sem+0x17e/0x1b8
  [c0407406] do_IRQ+0xa5/0xae
  [f8994311] t4_dacs+0x211/0x24b [wct4xxp]
  [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel]
  [c0457600] mempool_alloc+0x28/0xc9
  [c04ddd33] cfq_resort_rr_list+0x23/0x8b
  [c04deb6c] cfq_add_crq_rb+0xba/0xc3
  [c04dec72] cfq_insert_request+0x42/0x498
  [c04d5175] elv_insert+0x10a/0x1ad
  [c04d908b] __make_request+0x31d/0x366
  [c04de8b1] cfq_dispatch_requests+0x26a/0x46b
  [c04dde27] __cfq_slice_expired+0x8c/0xa5
  [c04de8b1] cfq_dispatch_requests+0x26a/0x46b
  [c04d505d] elv_next_request+0x15c/0x16a
  [f88bc101] start_io+0x77/0xdc [cciss]
  [f88bf63e] do_cciss_request+0x32c/0x337 [cciss]
  [f88ccff0] __split_bio+0x408/0x418 [dm_mod]
  [f88cd6a6] dm_request+0xce/0xd4 [dm_mod]
  [c04d6a81] generic_make_request+0x248/0x258
  [c04d8734] submit_bio+0xbf/0xc5
  [c04548e2] find_get_page+0x18/0x38
  [c04719ad] __find_get_block_slow+0xfb/0x105
  [c0471cea] __find_get_block+0x15c/0x166
  [c0471cea] __find_get_block+0x15c/0x166
  [c0471d24] __getblk+0x30/0x270
  [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd]
  [f885a472] journal_cancel_revoke+0x77/0x96 [jbd]
  [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd]
  [c041f871] __wake_up+0x2a/0x3d
  [f8856679] journal_stop+0x1b0/0x1ba [jbd]
  [c042a209] current_fs_time+0x4a/0x55
  [c048626d] touch_atime+0x60/0x8f
  [c04552ee] do_generic_mapping_read+0x421/0x468
  [c045478b] file_read_actor+0x0/0xd1
  [c04548e2] find_get_page+0x18/0x38
  [c0457319] filemap_nopage+0x192/0x315
  [c046048f] __handle_mm_fault+0x85e/0x87b
  [c047f46b] do_ioctl+0x47/0x5d
  [c047f6cb] vfs_ioctl+0x24a/0x25c
  [c047f725] sys_ioctl+0x48/0x5f
  [c0404eff] syscall_call+0x7/0xb
  ===
 VPM450: hardware DTMF disabled.
 VPM450: Present and operational servicing 2 span(s)
 Completed startup!
 About to enter startup!
 TE2XXP: Span 2 configured for CCS/HDB3/CRC4
 wct2xxp: Setting yellow alarm on span 2
 timing source auto card 0!
 SPAN 2: Secondary Sync Source
 Completed startup!


   Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy !

   For completeness sake, driver was previously loaded ok:

 Zapata Telephony Interface Registered on major 196
 Zaptel Version: 1.4.10
 Zaptel Echo Canceller: MG2
 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98
 Found TE2XXP at base address fdff, remapped to f8854000
 TE2XXP version c01a016a, burst ON
 Octasic optimized!
 FALC version: 0005, Board ID: 00
 Reg 0: 0x375a2400
 Reg 1: 0x375a2000
 Reg 2: 0x
 Reg 3: 0x
 Reg 4: 0x3101
 Reg 5: 0x
 Reg 6: 0xc01a016a
 Reg 7: 0x1300
 Reg 8: 0x
 Reg 9: 0x00ff2031
 Reg 10: 0x004a
 TE2XXP: Launching card: 0
 TE2XXP: Setting up global serial parameters
 Found a Wildcard: Wildcard TE220 (4th Gen)


   After trying lot's of things (disable ILO, disable USBs, try different 
 kernel,
   different TE220B, etc), I figured that this soft hangup does not show
   under zaptel 1.4.9.2...

   In all due honesty, I haven't got the faintest idea what kind of impact this
   could have.

   Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly
   a PC), the error does not show up as well.


   I checked the zaptel 1.4.10 ChangeLog

Re: [asterisk-users] Is Asterisk really good??

Quote 

We had a master source location.with a master image
 We cloned the hard drive with linux  dd copy of master image

Did the dd to clone it actually work on RAID devices 


Mike Trest - On Travel wrote:
 -Original Message-
 
 I'd be interested in sections like Rolling out a new server or How we
 maintain all the little configuration files without losing our sanity.
 

 Hi,

 I will contribute my 2-cents on how I maintained consistency on  a 
 large application
 with 64 +  Asterisks that all had to have the same config and links back to
 a central DB.

 Whenever we needed a new machine, we just

  We had a master source location.with a master image
  We cloned the hard drive with linux  dd copy of master image
  boot the new machine with this disk
  assign appropriate IP address
  perform some sanity checks prior to shipping
  Send either disk or full machine to remote COLO for physical install.

 After the machine came on line, it would have enough configuration to
 join the other members of the farm of asterisks.

 For intermediate updates, we used SSL-DSA keys between the master
 master image machine and each of the 64+ remotes.  We would wrote
 our own script and gave it a list of each machine on which to perform
 the particular steps.  When it was launched, we just went out to lunch
 or home at night while the remotes were updated.

 This application had as many as 6,000 simultaneous call running and
 we wrote the scripts such that each remote were placed in a
 take no calls status by the script so we did not kill any active traffic.

 We found that no canned package was useful to do this because each
 maintenance cycle was addressing a different part of the overall configuration
 and had slightly different commands that were needed.

 Any good script writer can do the same for what you described.

 Regards,  ..mike..



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Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?

exvito - I know it is a pain in the cahoonkus - but would you consider 
sharing the OTHER Digium board issues you are having , the recommended 
steps you were given by Digium to troubleshoot them, and the results ?
I think this real-wold experience wold be invaluable to the list.
THX in Advance for sharing !

Ex Vito wrote:
  Your stack trace appears to possibly be stack corruption.

  Could you try either this branch:
  http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/

  Or with a kernel that does not have 4K stacks enabled?  You can check if 
 your installed kernel does with the following command.

  $ cat /boot/config-`uname -r` | grep 4K
  # CONFIG_4KSTACKS is not set

 

   ...thanks for your feedback Shaun.

   I am currently nearing other troubleshooting issues regarding
   a TC400B (which will probably lead me to get in touch with
   Digium install support).

   So I have no schedule today to test your suggestions; maybe
   tomorrow / thursday.

   They are noted, however. :)

   Cheers,
 --
  exvito

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Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?

Shaun - Could you clarify your post a bit ?

1 - Is the 4 K  stacks a Known Problem ?
 a) If so is it known to be problem on any specific Linux distro ?
 b) Should ALL installation Check for this PRIOR to doing an 
Asterisk Install ?

2) The branch you mention below - are fixes from it in Any current * 
release ?


Shaun Ruffell wrote:
 Your stack trace appears to possibly be stack corruption. 

 Could you try either this branch:
 http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/

 Or with a kernel that does not have 4K stacks enabled?  You can check if your 
 installed kernel does with the following command.

 $ cat /boot/config-`uname -r` | grep 4K
 # CONFIG_4KSTACKS is not set

 Cheers,
 Shaun


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Re: [asterisk-users] Is Asterisk really good??

2008-04-14 Thread Al Baker

why yes, my rsync does that just fine, you must not be running
the latest version

Steve Edwards wrote:
 On Mon, 14 Apr 2008, Bernd Felsche wrote:

   
 Steve Edwards [EMAIL PROTECTED] wrote:

 
 I'm mainly interested in consistency in configuration. The method has
 to be sophisticated enough to handle this box has 2 Ethernet interfaces
 so I should configure OpenSER and Asterisk to listen to both IP addresses
 on ports 5060 and 5061 respectively. This would preclude rsync.
   
 Why do you think that that would preclude rsync?
 

 Well, it may be based on my ignorance :)

 Can rsync mung a stanza from iax.conf like:

 [general]
   disallow   = all
  allow   = ulaw
  mailboxdetail   = no
  notransfer  = yes
  port= 5036
  register= ${HOSTNAME}:[EMAIL PROTECTED]
  trunk   = no

 and insert the appropriate values?

 Can rsync create /etc/sysconfig/openser like:

 # Created by ./host-setup.sh on 2008-04-12 17:55:03

  OPTIONS=
  OPTIONS=$OPTIONS -l a.b.c.d:5060
  OPTIONS=$OPTIONS -l a.b.c.e:5060

 # (end of /etc/sysconfig/openser)

 where a.b.c.d and a.b.c.e are the IP address of eth0 and eth1? (And the 
 3rd line would only be created if there are 2 interfaces.)

 Thanks in advance,
 
 Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Is Asterisk really good??

Remember - True TELCO grade systems simply cannot be compared to 
anything else.
You want the reliability , uptime, and all the bells and whistles of 
true Carrier Grade Hardware/Software
then you pay for it. If you want something you can tinker under the 
hood with and do some cool
stuff with, Asterisk is it. BUT - by it need  NORTEL PBX reliability or 
the power and flexibility
of a big Rockwell ACD, get out your check book. The two are not the same.

Eugen Soare wrote:
 So this is just a general question, Is Asterisk really good?

 Reliability?

 Functionality?

 Customization's?


 I am coming from a Nortel world, were you pay for everything, and you 
 can't delve into the software. But it seems that customization would be 
 a great thing.
 Like, setting up a war-dialer to customer lists, incoming/outgoing 

   

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Re: [asterisk-users] DTMF between Asterisk servers.

Just a thought. A while back there was discussion about the merits of
having a product (in that case an O/S) with contracted vendor support
or relying solely on list support.
I note in the post below where one responder states
 It may also have been because less than 23 hours had elapsed

Different strokes for different folks But  23 Hours is lng 
time in
the production world with no help on a TELCO problem. Just an observation on
how differently folks see things and what folk need to recognize before
they dump their NORTEL etc and jump into open-source.

Mark Hamilton wrote:
 No, I tried calling the inbound DID to see if DTMF passes through. And most
 times it does, however, it's not being relayed to the Asterisk server 2, and
 then to the direct external phoneline.

 I tried changing all dtmfmodes for the sip peer, for the inbound DID
 provider, and it didn't work, even tried playing with canreinvite, etc.

 Hence why my desperate plea for help.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
 Sent: April 8, 2008 11:03 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] DTMF between Asterisk servers.

 I believe that what you described should just work with the caveat
 that dtmf=inband is rarely the right thing to do over SIP, and is
 prone to all sorts of DTMF detection and debounce issues.

 I assume you've tried calling a POTS endpoint and listening to see if
 you get DTMF passed through?

 1) You did not give a great deal of information about what the current
 situation was, or what investigations you've already tried, which is
 probably why no-one felt they could reply.
 2) It may also have been because less than 23 hours had elapsed...

 Regards,
 Steve

 On 08/04/2008, Mark Hamilton [EMAIL PROTECTED] wrote:
   
 I find it  hard to believe no one knows, so is it just plain no helping? J

 If someone would like to atleast point me in the right direction that will
 deal specifically with what I'm asking, that would be appreciated too.

 Much thanks.

 From: Mark Hamilton [mailto:[EMAIL PROTECTED]
  Sent: April 7, 2008 11:48 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: DTMF between Asterisk servers.

 Hello,

 I'm a little confused on DTMF.

 A sip peer is registered on two Asterisk servers. No dtmfmode is set for
 them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
 register on each other.



 A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the
 
 call
   
 is transferred to Asterisk 2:


 
 RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/[EMAIL PROTECTED],,t
 T,)
   
 Where 12351 accepts the call on Asterisk 2, and in some cases, that call
 
 is
   
 transferred out to a PSTN number, or wherever, but not within Asterisk
 anymore via provider2, dtmf=rfc2833.

 When the call comes in, I'd like it to relay DTMF just dandy. How can I do
 so?

 There is no NAT between the Asterisk servers or in front of them. However,
 Asterisk2 has iptables which allows all UDP traffic  to/fro Asterisk1.
 
 When
   
 Asterisk2 transfers the call to external endpoints, there might be a LAN,
 but relative ports are open on those LANs.

 Please help.

 Thanks in advance,

 Mark.
 

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Re: [asterisk-users] Is Asterisk really good??

Quote

And for what it's worth, at my new job I babysit about a 16-machine
cluster running VICIdial for close to 200 agents, and by and large, it
just runs.  It's got about 20 T-1s feeding it

Please share more about this.

What/How are you clustering the boxes ?

Is this all VOIP  or TDMF front and VOIP for agents in back ?

What kind of Boxes ?   What O/S

What tools are you using to monitor this big-azz mother ?

Thx in in advance for sharing !!

Jay R. Ashworth wrote:
 On Thu, Apr 10, 2008 at 04:05:47PM -0400, Al Baker wrote:
   
 Remember - True TELCO grade systems simply cannot be compared to
 anything else. You want the reliability , uptime, and all the bells
 and whistles of true Carrier Grade Hardware/Software then you pay for
 it. If you want something you can tinker under the hood with and
 do some cool stuff with, Asterisk is it. BUT - by it need NORTEL PBX
 reliability or the power and flexibility of a big Rockwell ACD, get
 out your check book. The two are not the same.
 

 Sure.  But even commercial PBX/Hybrid stuff is often more robust than
 Asterisk-on-a-PC (I don't have enough anecdotes on appliances, though I
 expect they'd be better).

 Now, yes, carrier-grade equipment can be as complex and still 4-nines
 reliable (or 5-nines :-), but we're now talking NEBS, -48VDC, 23-inch
 racks, and 6-digit-plus price tags.

 And for what it's worth, at my new job I babysit about a 16-machine
 cluster running VICIdial for close to 200 agents, and by and large, it
 just runs.  It's got about 20 T-1s feeding it.

 Cheers,
 -- jra
   

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[asterisk-users] Excellent Paper on ECHO in VOIP Environment


http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html

I thought others would find this helpful.

I have no tie to any vendor or company referenced in this paper and I 
believe the same concepts to be applicable to the Asterisk world.
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[asterisk-users] Any body tried MysqlPool-1.4

I saw an add-on for Asterisk at 
http://www.yosd.at/index.php?option=com_contenttask=viewid=23Itemid=38

Anyone tried any of these ?

How well did they work ??


Asterisk Applications   Print 
http://www.yosd.at/index2.php?option=com_contenttask=viewid=23pop=1page=0Itemid=38
 


The Asterisk PBX core system offers you great flexibility because of its 
internal design. It does come with support for a varity of codecs, 
channels and applications. But sometimes there is the need for a special 
application which is not included in the basic system...

Here you can find some very usefull asterisk addons, which we have 
developed in the past time. Download them - use them. If you need 
support setting up one of the applications, or if you need a special 
version (with custom modifications) then contact us under This email 
address is being protected from spam bots, you need Javascript enabled 
to view it . We do also offer creating complete new applications for the 
asterisk pbx system.

 

* MysqlPool

MysqlPool is not an asterisk application. MysqlPool is an asterisk
ressource which is needed for the following applications to work
correctly. The MysqlPool ressource will handle multiple connections
to one or more Mysql servers using sockets or tcp/ip. It does
register some CLI commands which makes it possible to get the state
of the connections, and to change to number of connections for one
host. The ressource does offer public functions for the other
applications to get a valid Mysql connection handle without locking
problems.

* LCDial

LCDial is a failsafe least cost routing engine.

* DBQuery

With DBQuery you can access you Mysql Database and query it for the
needed data, or update data in your Mysql Database. It is also
possible to register DBQuery as cdr backend - so you can define your
own sql statement for writing cdr data.

* DBRewrite

DBRewrite offers you regular expression matching and substitution.
It does support multiple keys, and for each key you can define as
many regular expessions and substitution rules as you want.




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Re: [asterisk-users] Where is the Digium DS3 card?

2008-04-06 Thread Al Baker

ok - but has a new release date be announced  ?
or
Has Digium officially dropped the product ?

Steve Totaro wrote:
 On Sun, Apr 6, 2008 at 3:47 AM, Tilghman Lesher
 [EMAIL PROTECTED] wrote:
   
 On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
   Any know what Digium hasn't released the DS3 card?
  
   It was supposed to be out a while ago.

  There was a fundamental problem with the chipset used, which precluded the
  card from being useful.  Specifically, the chipset only permitted the first
  255 channels to be addressed (instead of the full 672).  Since that time, 
 and
  partly due to this circumstance, Digium no longer announces the release of
  cards until they are ready to be shipped, with drivers and all.

  --
  Tilghman

 

 Sounds prudent.  I liken it to announcing a vehicle that runs on
 water, maybe, one day.

 Anyways, who or why would anyone want to put all 28 eggs in one basket?

 Thanks,
 Steve Totaro

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Re: [asterisk-users] Asterisk with lumenvox

2008-04-06 Thread Al Baker

I don't use Lumenvox, yet.
I have a large client who has expressed interest in having something 
like it deployed.
For such a revolutionary product there is very little chatter in the 
list about it.
I really had hoped to hear from a bunch of people who had deployed it 
and how it worked out for them.

Josué Conti wrote:
 Hi Al, how are you?
 You use Lumenvox? What we think of the performance of the engine?
 Thank you for your attention

 Regards!

 Josué

 2008/4/5, Al Baker [EMAIL PROTECTED]:
   
 I had posted earlier asking about folks real world experiences with
 with Lumenvox, and the thread 'strangely' disappeared after some
 bloke from down under justed sodded himself over my straight simple
 questions.
 Hm- makes you wonder.

 Josué Conti wrote:
 
 Hello everyone.
 I wish I could continue with the approval of the engine Lumenvox, for
 voice recognition, but not a development of acceptable engine,
 Please could help me in achieving test?
 As I said earlier we have a project that will involve a very large
 number of licenses for Voice recognition, but I would count on help
 from Lumenvox, for this case.
 Could you help me?

 Best Regards

 Josué

 2008/3/19 Josué Conti [EMAIL PROTECTED]:

   
 Hello everyone, Rodrigo and Philipp Hello, I would like to know how to
  properly configure the engine Lumenvox no asterisk, I am trying to
  dial by vox actually like that the user should dial for receipt of my
  business, is attended by an IVR system with voice recognition that
  allows the user to say who would like to talk and the asterisk foward
  the call.
  Set up the asterisk below, but the system recognizes the voice, but
  does not guide the call, running immediately after a hangup, what is
  wrong with my settings? I can not very material support on the issue,
  could help me?
  I am not really achieving great results in my tests with engine Lumenvox:
  I am trying to test a simple scheduling dialing by voice, where the
  system identify the user by name and system called in your phone
  number, but I am not able, could help me?
  If I did not say any word, the system is static, but if I say any
  Word, even different words grammar.gram (ura.gram) of the system
  Performs the following priorities file extensions.conf, please, can
  You help me?

  Best Regards

  Josué

  Our programming files are configured this way:
  Ipbx: / etc / asterisk # vim lumenvox.conf
  ; LumenVox configuration file
  [General]
  Servers = 127.0.0.1; Speech Engine Servers to use.
  Save_sound_files = no; Set to yes to save sound files for use with Speech 
 Tuner
  [Grammars]
  ura = / etc / asterisk / grammars / ura.gram
  [Default]
  Vad_snr_sensitivity = 50
  Vad_volume_sensitivity = 50
  Vad_eos_delay = 1250
  Vad_wind_back = 750
  End_of_speech_timeout = 15000
  Use_oov_filter = no
  ;;
  ;; 
  Ipbx: / etc / asterisk # vim extensions.conf
  [General]

  [Globals]

  DYNAMIC_FEATURES = # pickupexten hangup atxfer # # blidxfer

  [Default]
  Length = 2000.1, Playback (Ura / instit / instit_casa)
  Length = 1515.1, Playback (Ura / parabens)

  ;;
  ;;
  ; Pilot URA
  Length = 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1)
  Length = 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1)
  Length = 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1)
  Length = 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1)

  ; IVR URA

  ;
  [URA]
  ;
  Length = s, 1, Answer ()
  Length = s, n, Wait (3)
  Length = s, n, NoOp (entry Ura)
  Length = s, n, Set (TRIES = 0)
  ; Length = s, n, ResponseTimeout (10)
  Length = s, n, BackGround (Ura /abertura)
  Length = s, n, Playback (beep)
  ; Length = s, n, BackGround (Ura / abertura1)
  Length = s, n, Goto (lumenvox-test, s, 1)
  [Lumenvox-test]
  Length = s, 1, Answer
  Length = s, n, Wait (1)
  Length = s, n, SpeechCreate ()
  Length = s, n, SpeechActivateGrammar (Ura)
  Length = s, n, SpeechStart ()
  Length = s, n, SpeechBackground (liggol / abertura)
  Length = s, n, SpeechDeactivateGrammar (Ura)
  Length = s, n, Goto (institutional, s, 1 - $ SPEECH_TEXT (0) ())
  [Institutional]
  Length = s, 1, Playback (Ura / instit / instit)
  Length = s, 2, congestion (3)
  Length = s, 3, hangup

  ipbx: / etc / asterisk / grammars # vim ura.gram
  # - Grammar: ura.gram


  # ABNF 1.0;
  Language es-CO;
  Voice mode;
  Tag-format lumenvox/1.0;
  Root $ URA;
  $ Continent = ((Josue | Conti) []): 2000;
  $ Palms = () 1515 ;

  $ Ura = ($ conti | $ palms) = $ $ $ ();

  2008/3/19, Rodrigo Gonzalez [EMAIL PROTECTED]:



 
 Josué Conti escribió:

   
Hello all, how are you?
I would like to know from someone uses or has used the engines of
LumenVox for integration

Re: [asterisk-users] Asterisk with lumenvox

2008-04-05 Thread Al Baker

I had posted earlier asking about folks real world experiences with
with Lumenvox, and the thread 'strangely' disappeared after some
bloke from down under justed sodded himself over my straight simple 
questions.
Hm- makes you wonder.

Josué Conti wrote:
 Hello everyone.
 I wish I could continue with the approval of the engine Lumenvox, for
 voice recognition, but not a development of acceptable engine,
 Please could help me in achieving test?
 As I said earlier we have a project that will involve a very large
 number of licenses for Voice recognition, but I would count on help
 from Lumenvox, for this case.
 Could you help me?

 Best Regards

 Josué

 2008/3/19 Josué Conti [EMAIL PROTECTED]:
   
 Hello everyone, Rodrigo and Philipp Hello, I would like to know how to
  properly configure the engine Lumenvox no asterisk, I am trying to
  dial by vox actually like that the user should dial for receipt of my
  business, is attended by an IVR system with voice recognition that
  allows the user to say who would like to talk and the asterisk foward
  the call.
  Set up the asterisk below, but the system recognizes the voice, but
  does not guide the call, running immediately after a hangup, what is
  wrong with my settings? I can not very material support on the issue,
  could help me?
  I am not really achieving great results in my tests with engine Lumenvox:
  I am trying to test a simple scheduling dialing by voice, where the
  system identify the user by name and system called in your phone
  number, but I am not able, could help me?
  If I did not say any word, the system is static, but if I say any
  Word, even different words grammar.gram (ura.gram) of the system
  Performs the following priorities file extensions.conf, please, can
  You help me?

  Best Regards

  Josué

  Our programming files are configured this way:
  Ipbx: / etc / asterisk # vim lumenvox.conf
  ; LumenVox configuration file
  [General]
  Servers = 127.0.0.1; Speech Engine Servers to use.
  Save_sound_files = no; Set to yes to save sound files for use with Speech 
 Tuner
  [Grammars]
  ura = / etc / asterisk / grammars / ura.gram
  [Default]
  Vad_snr_sensitivity = 50
  Vad_volume_sensitivity = 50
  Vad_eos_delay = 1250
  Vad_wind_back = 750
  End_of_speech_timeout = 15000
  Use_oov_filter = no
  ;;
  ;; 
  Ipbx: / etc / asterisk # vim extensions.conf
  [General]

  [Globals]

  DYNAMIC_FEATURES = # pickupexten hangup atxfer # # blidxfer

  [Default]
  Length = 2000.1, Playback (Ura / instit / instit_casa)
  Length = 1515.1, Playback (Ura / parabens)

  ;;
  ;;
  ; Pilot URA
  Length = 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1)
  Length = 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1)
  Length = 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1)
  Length = 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1)

  ; IVR URA

  ;
  [URA]
  ;
  Length = s, 1, Answer ()
  Length = s, n, Wait (3)
  Length = s, n, NoOp (entry Ura)
  Length = s, n, Set (TRIES = 0)
  ; Length = s, n, ResponseTimeout (10)
  Length = s, n, BackGround (Ura /abertura)
  Length = s, n, Playback (beep)
  ; Length = s, n, BackGround (Ura / abertura1)
  Length = s, n, Goto (lumenvox-test, s, 1)
  [Lumenvox-test]
  Length = s, 1, Answer
  Length = s, n, Wait (1)
  Length = s, n, SpeechCreate ()
  Length = s, n, SpeechActivateGrammar (Ura)
  Length = s, n, SpeechStart ()
  Length = s, n, SpeechBackground (liggol / abertura)
  Length = s, n, SpeechDeactivateGrammar (Ura)
  Length = s, n, Goto (institutional, s, 1 - $ SPEECH_TEXT (0) ())
  [Institutional]
  Length = s, 1, Playback (Ura / instit / instit)
  Length = s, 2, congestion (3)
  Length = s, 3, hangup

  ipbx: / etc / asterisk / grammars # vim ura.gram
  # - Grammar: ura.gram


  # ABNF 1.0;
  Language es-CO;
  Voice mode;
  Tag-format lumenvox/1.0;
  Root $ URA;
  $ Continent = ((Josue | Conti) []): 2000;
  $ Palms = () 1515 ;

  $ Ura = ($ conti | $ palms) = $ $ $ ();

  2008/3/19, Rodrigo Gonzalez [EMAIL PROTECTED]:


 
 Josué Conti escribió:
   
Hello all, how are you?
I would like to know from someone uses or has used the engines of
LumenVox for integration with the asterisk for voice recognition.
   
Best Regards
   
Josué
   
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Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk withlumenvox

2008-04-03 Thread Al Baker

Quote Re-reading your answer it appears you provided advice about a topic
where you hadn't actually utilized the product.

No - I asked a question  to find out peoples experience.
I did not Offer advice to anybody, I ASKED for advice vis-a-vis where They had 
actually
used the product.


And yes - if anyone wishes to share more about their experiences I'd like to 
hear it.



Dean Collins wrote:
 Ok, I think that original request was answered more than a few days ago
 by people who had actually used Lumenvox.

 Re-reading your answer it appears you provided advice about a topic
 where you hadn't actually utilized the product.

 As several people who answered Phillip's original question - Lumenvox is
 a good product and at an affordable price point.

 Just understand it is utterance recognition with limited response set
 recognition - NOT NLVR.

  

 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED] 
 +1-212-203-4357
 +61-2-9016-5642 (Sydney in-dial). 


   
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Thursday, 3 April 2008 12:14 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk
 withlumenvox

 Thank you.
 I think that part of one's research is to ask in a user group what
 peoples real word experience of a product has been. Particularly
 
 since
   
 I was responding to a post by Philip in which he had said
   So what is that you'd like to know?
Philipp

 in response to  a question of

I would like to know from someone uses or has used the engines of
LumenVox for integration with the asterisk for voice recognition.

 posted by another member of the mailing list.

 Sorry you don't agree.

 However, I would still be interested in hearing anyone experiences
 
 that would care
   
 to share them.

 And to those kind enough to share I say 'thanks' in advance.

 g
 Dean Collins wrote:
 
 Hi Al,

 I'm saying this politely so don't take it the wrong way. Go away and
   
 do
   
 some research.

 Learn the difference between NLVR (Dragon Dictate) and limited set
 utterance recognition (Lumenvox).

 Lumenvox is a great product for the price point and as their
   
 developer
   
 kit is so reasonable you should buy one to have a play on a test
   
 bench
   
 with as a basic minimum.



 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED]
 +1-212-203-4357
 +61-2-9016-5642 (Sydney in-dial).



   
 -Original Message-
 From: [EMAIL PROTECTED]
 
 [mailto:asterisk-users-
   
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Wednesday, 2 April 2008 11:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk

 
 withlumenvox

   
 In the real world, just how good of recognition can you get based
 
 on
   
 your experience ?

 How much processing power do you find it takes ? I know that a

 
 dedicated

   
 Voice Recognition for the
 PC such as Dragon Naturally Speaking requires :
 -a pretty beefy system
 -that you use a limited set of microphones
 - and ideally, a 20 session to train the software.

 Clearly all of this not feasible in a IVR environment, so, in the
 absence of all this, just how good , and how sophisticated of a
 
 voice
   
 recognition can one achieve ?

 Philipp von Klitzing wrote:

 
 Hi!



   
 I would like to know from someone uses or has used the engines of
 LumenVox for integration with the asterisk for voice recognition.


 
 So what is that you'd like to know?

 Philipp


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Re: [asterisk-users] Problems with DELL 1600

2008-04-02 Thread Al Baker

Could you be like 1% more specific, with perhaps including any 
relevant Errors, Log Files ?

What makes you think it is the DELL , not your T1 boards, or your 
service provider  or 

Ruben Zamora wrote:
 Hi

 I just want to know if anyone have problems with server DELL 1600, 
 Like:  Hangup Call.

 Thanks

 Ruben

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[asterisk-users] IVR Asterisk Voice Recognition - Asterisk with lumenvox

2008-04-02 Thread Al Baker

In the real world, just how good of recognition can you get based on 
your experience ?

How much processing power do you find it takes ? I know that a dedicated 
Voice Recognition for the
PC such as Dragon Naturally Speaking requires :
-a pretty beefy system
-that you use a limited set of microphones
- and ideally, a 20 session to train the software.

Clearly all of this not feasible in a IVR environment, so, in the 
absence of all this, just how good , and how sophisticated of a voice 
recognition can one achieve ?

Philipp von Klitzing wrote:
 Hi!

   
 I would like to know from someone uses or has used the engines of
 LumenVox for integration with the asterisk for voice recognition. 
 

 So what is that you'd like to know?

 Philipp


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Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk withlumenvox

2008-04-02 Thread Al Baker

Thank you.
I think that part of one's research is to ask in a user group what 
peoples real word experience of a product has been. Particularly since 
I was responding to a post by Philip in which he had said
  So what is that you'd like to know?
   Philipp

in response to  a question of 
  
   I would like to know from someone uses or has used the engines of
   LumenVox for integration with the asterisk for voice recognition.

posted by another member of the mailing list.

Sorry you don't agree.

However, I would still be interested in hearing anyone experiences that would 
care to share them.

And to those kind enough to share I say 'thanks' in advance.

g
Dean Collins wrote:
 Hi Al,

 I'm saying this politely so don't take it the wrong way. Go away and do
 some research.

 Learn the difference between NLVR (Dragon Dictate) and limited set
 utterance recognition (Lumenvox).

 Lumenvox is a great product for the price point and as their developer
 kit is so reasonable you should buy one to have a play on a test bench
 with as a basic minimum.

  

 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED] 
 +1-212-203-4357
 +61-2-9016-5642 (Sydney in-dial). 


   
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Wednesday, 2 April 2008 11:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk
 
 withlumenvox
   
 In the real world, just how good of recognition can you get based on
 your experience ?

 How much processing power do you find it takes ? I know that a
 
 dedicated
   
 Voice Recognition for the
 PC such as Dragon Naturally Speaking requires :
 -a pretty beefy system
 -that you use a limited set of microphones
 - and ideally, a 20 session to train the software.

 Clearly all of this not feasible in a IVR environment, so, in the
 absence of all this, just how good , and how sophisticated of a voice
 recognition can one achieve ?

 Philipp von Klitzing wrote:
 
 Hi!


   
 I would like to know from someone uses or has used the engines of
 LumenVox for integration with the asterisk for voice recognition.

 
 So what is that you'd like to know?

 Philipp


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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards

2008-03-31 Thread Al Baker

Thank you for all your time om your most detailed response.
It is extremely helpful.

The vendor's web page is
http://www.penguincomputing.com/index.php?option=com_contentid=170Itemid=209task=viewsysid=10007609

*PCI EXPANSION SLOTS*
Number of Slots 5
Slot Speed PCI Express: two x8 slots, two x8 low profile slots;
PCI-X: 64-bit/100MHz

or if that doesn't display

*PCI EXPANSION SLOTS*
Number of Slots 5
Slot SpeedPCI Express: two x8 slots, two x8 low profile slots;
PCI-X: 64-bit/100MHz

or
PCI EXPANSION SLOTS
Number of Slots 5
Slot Speed PCI Express: two x8 slots, two x8 low profile slots;
PCI-X: 64-bit/100MHz

Nick Seraphin wrote:
On Sat, 29 Mar 2008, Al Baker wrote:

Detailed specs for the types of PCI slots on the system were posted
each and every time I posted int the line

Actually, your description wasn't 100% clear at all.

_PCI Express_*: _two x8 slots*_, _two x8 low profile slots*_; *_PCI-X:
64-bit/100MHz_*

1) This description seems to IMPLY that there are 5 slots total. Do you
know if this is in fact correct? It implies there are 2 PCI-E x8 slots, 2
PCI-E x8 low profile slots, and 1 PCI-X slot. I wouldn't rely on that
however without talking to the vendor.

2) The first PCI Express: heading would normally imply that the slots
listed afterwards are ALL PCI Express slots, however PCI-X is not PCI
Express, so the vendor's description is confusing and misleading.

3) All these damn *'s you keep inserting, are those all done by you, or
are some of them from the web page description? Most of the time when
something has a * by it that means it's conditional on a footnote that
appears at the bottom of the section or page. Are there footnotes we need
to know about to clarify this?

4) Is this a rackmount server or a tower case? If rackmount, is it a 1u
server or a 2u server or a 4u server? Just because the motherboard has 5
slots doesn't mean the case it is installed in will support 5 cards. A 1u
case rarely supports more than 1 or 2 cards, and always requires a riser
card. A 2u server rarely supports more than 2 cards unless they are
low-profile. A 4u server might allow 5 cards, IF the case is designed
with 5 slot openings in the back.

5) Are all the card slots open and available to you at time of shipping?
Many options a customer orders with a server, such as a RAID controller or
additional network ports will fill one or more of the available slots.
You need to be sure all the slots you need are available to you when you
get the server.

As for types of cards. As others have already said, PCI-X is not
PCI-Express and they are not interchangeable.

A PCI-Express card slot can accomodate any PCI-Express card with the same
number of lanes or less. So an x8 slot (8 lanes) will support an x1, x2,
x4, or x8 card, but not an x16 card. I believe the Digium PCI Express
cards are only x1 (one lane) so they should fit in any PCI Express slot,
but you should check with Digium's web site to be 100% sure the card you
are buying is a x1 card.

Unless you specifically buy a low-pofile card, a normal PCI or PCI Express
Card will NOT fit in a low-profile slot. So assuming Digium's cards are
full height, you only have 3 possible options. The 2 PCI Express full
height slots, and the 1 PCI-X slot, assuming all those slots are open and
will be available with the case you're using.

I would NOT base my purchasing decision on that vague description given by
the vendor that you have listed in your messages. I would contact the
vendor and clarify the total number of slots, types of slots, whether they
are open or not, and whether the case will support them all.

Many vendors will use a motherboard with 3-5 slots on the board, in a 1u
rackmount case that only supports 1 physical card.

One final word of warning... don't try to stick too many cards in one box
without double checking with someone who can tell you if it will handle
that capacity or not. There were a lot of problems with early Digium
4-port T1 cards where you couldn't use more than 1 or 2 cards at a time
because of the interrupts. The newer cards, especially PCI Express, may
not have that problem anymore... but I would double check before
proceeding with more than 2 cards in one box.

Can anyone out there clarify (for me as well) whether you can put, say, 4
or 5 4-port T1 cards in a single box now and have it work ok? Assuming
enough RAM and a fast enough CPU of course.

-- Nick

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Re: [asterisk-users] Had it with Dell Garbage - HP Question

2008-03-30 Thread Al Baker

Could you elaborate a bit more on :
For example, if I install zaptel from source, your support contract 
with them is void.

Does this mean it is impossible to run Asterisk on Vendor Supported 
versions of RedHat or Suse ?

Thanks

Michiel van Baak wrote:
 On 02:34, Sat 29 Mar 08, Al Baker wrote:
   
 Helps a bunch !!!
 One follow up question - out of all of your possible choices for the OS 
 how did you pick *Debian*.
 I 'm not saying is bad, I just know nothing about the particular disto. 
 and and very curious what
 it brought to the table that made you pick over say *RedHat* - where you 
 can *buy support *or *SUSE* - where you can *buy support*. My fear from 
 hell is that I' get 50 or 60 of these boxes in, start having kernel 
 panics, and have no damn body to help except the folks on mailing lists. 
 Mind you these are often really smart people, very generously giving of 
 their time, but not quite the say as a manned/paid support organization.
 

 I choose Debian because I was already using it.
 And because there are people out there that can help me.

 I dont want the support from suse or redhat because they
 wont help me when running anything that's not in their
 repositories.

 For example, if I install zaptel from source, your support
 contract with them is void.

 I also really like the Open and Free mindset of Debian.

   


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Re: [asterisk-users] Had it with Dell Garbage - HP Question

2008-03-30 Thread Al Baker

There are people who will support your Debian / Centos / whatever boxes.

If it is OK to ask on a non-commercial list, do you have a list of 
reliable O/S support folks.
By this I mean companies with a support staff, as opposed to a really 
bright and talented guy
who does it between classes in school.
Historically our projects were on big HP iron with HP-UX support from HP

THX

Tzafrir Cohen wrote:
 On Sat, Mar 29, 2008 at 02:34:36AM -0400, Al Baker wrote:
   
 Helps a bunch !!!
 One follow up question - out of all of your possible choices for the OS 
 how did you pick *Debian*.
 I 'm not saying is bad, I just know nothing about the particular disto. 
 and and very curious what
 it brought to the table that made you pick over say *RedHat* - where you 
 can *buy support *or *SUSE* - where you can *buy support*. My fear from 
 hell is that I' get 50 or 60 of these boxes in, start having kernel 
 panics, and have no damn body to help except the folks on mailing lists. 
 Mind you these are often really smart people, very generously giving of 
 their time, but not quite the say as a manned/paid support organization.
 

 What exactly is supported?

 Specifically, RHEL does not include Zaptel. And is not likely to include
 the kernel Zaptel modules until Zaptel comes closer to mainline kernel.

 SLES includes a Zaptel package of its own. 1.2.4 .

 Will they support a system that has unsupported kernel code?


 What is the alternative? buy support elsewhere. There are people who
 will support your Debian / Centos / whatever boxes. With RHEL and SuSE
 you have to buy support. With Debian it is optional.

   

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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards

Detailed specs for the types of PCI slots on the system were posted 
each and every time I posted int the line

I an considering using *your High Density T*1 cards on a number of 
servers we are considering purchasing. The _VENDOR lists_ that his 
_SYSTEM has:
_
_PCI Express_*: _two x8 slots*_, _two x8 low profile slots*_; *_PCI-X: 
64-bit/100MHz_* 

I just do not know how to ask any more clearly, but as always I am open 
to suggestions.

*_To Jared Smith  - THANK YOU VERY MUCH 
_*

Steve Totaro wrote:
 Are you looking to purchase a server or just looking for a card for an
 existing server?

 Either way, post the specs from the manufacturer (and also make sure
 those slots are open)

 You will get your answer pretty quickly, and much less painfully this way.

 Thanks,
 Steve Totaro

 On Thu, Mar 27, 2008 at 2:30 PM, Jared Smith [EMAIL PROTECTED] wrote:
   
 On Thu, 2008-03-27 at 12:16 -0400, Al Baker wrote:
   ==
   Here is the Confusion   - 1)  Can ANY of your QUAD cards go in the PCI 
 Express - two x8 slots   - Yes or NO ?

  Yes, the *TE420B *will fit.  It's a 4-port T1/E1/J1/PRI card with echo
  cancellation on-board.  It only fits in PCI Express slots.


  
  2  Can ANY of your QUAD cards go in the PCI 
 Express - two x8 LOW Profile Slots  - Yes or No ?

  No, none of our PCI Express cards are low-profile (as far as I know)


  3) Can your Quad cards you list as fitting a 
 PCI or PCI-X 3.3 Volt Slots  fit in  the PCI-X: 64-bit/100MHz slot
 OR would I have to ask the vendor Is this 
 PCI-X:  64-bit/100MHz  definitely a 3.3 Volt Slot ?
  Yes - It would absolutely definitely fit 
 OR NO - It will NOT fit   OR   To answer that I need more info  ?
  

  *You would need to ask your vendor whether or not the PCI-X slot is a 3.3
  volt or a 5.0 volt slot.  I'm guessing it's 3.3 volts, but that's just a
  guess.*

  Here's a link to a decent article explaining the difference between PCI
  and PCI Express slots, if you're interested.
  http://www.geeks.com/techtips/2006/techtips-24sept06.htm


  --
  Jared Smith
  Community Relations Manager
  Digium, Inc.


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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards

Thank you Brad .

Watkins, Bradley wrote:

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Thursday, March 27, 2008 12:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Question about PCI Slots for 
 DIGIUMs Boards

 No actually I gave it a LOT of thought and I and even asked two 
 different techs who repair PCs
 and both said what the to vendors are saying is not 
 sufficiently clear 
 that I would
 make a purchasing decision based on what you have in hand from them
 But, nice try at the cheap shot.

 The two techs you spoke with are obviously not familiar enough with the
 technology at hand to be able to make that determination, because all of
 the necessary information is contained within the original e-mail and
 the specs available on Digium's website.

 To simplify, however, here is the answer:  The TE420 cards will fit in
 the regular x8 (not low-profile) PCI-E slots, and the TE412P will fit
 into the PCI-X slots.

 Regards,
 - Brad

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Re: [asterisk-users] Had it with Dell Garbage - HP Question

Helps a bunch !!!
One follow up question - out of all of your possible choices for the OS 
how did you pick *Debian*.
I 'm not saying is bad, I just know nothing about the particular disto. 
and and very curious what
it brought to the table that made you pick over say *RedHat* - where you 
can *buy support *or *SUSE* - where you can *buy support*. My fear from 
hell is that I' get 50 or 60 of these boxes in, start having kernel 
panics, and have no damn body to help except the folks on mailing lists. 
Mind you these are often really smart people, very generously giving of 
their time, but not quite the say as a manned/paid support organization.

Thx for sharing !!!



Michiel van Baak wrote:
 On 08:02, Thu 27 Mar 08, Al Baker wrote:
   
 How do you get notifications ?
 Is this thru one of the add on packages HP sells for the box ?  Which One ?
 Could you be more specific about what you mean by a recovery CD
 and hod do you get console access below multi used to do recovery ??

 What is integrated ILO BIOS Access sounds cool.

 What O/S you usin and what made you pick it ?

 What kind and how many RAIDS are you using. The HP site gave like 8 
 different RAID controllers and like 20 CPUs to chose from.  How did you 
 chose ?

 Thx for sharing !!!
 

 I'm not the op, but sending a reply anyways.

 The notifications come from the HP tools you can download
 for free from their website.

 The recovery cd is probably a selfmade installer for their
 setup. At least that's what we have.

 the ILO stuff is to give you access to the box like you were
 sitting right in front of it with a physical keyboard and
 monitor, but over IP.
 You can boot the machine, access the cd in your local
 machine etc, even if the box is on the other side of the
 moon.

 We use Debian. HP even supports it on their DL380 boxen.

 We use the P400 raid controller. Setup RAID5 with 3 disks.
 CPU we use right now is the Intel E5405

 Hope this helps a bit.

   

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Re: [asterisk-users] Had it with Dell Garbage - HP Question

How did you chose Centos, versus Red Hat, Suse, Debian, ?
Was there some key feature it offered that the others didn't ?
Cost ?

Darren Wright wrote:
 Notifications can be done either thru SNMP traps or SMTP.  Insight
 Manager is free from HP, but any SNMP trapper can work with alerts.

 The recovery CD is just a build that reloads the majority of the system
 with a static ip.   We backup off site to one of our servers via FTP.

 ILO access is an integrated IP KVM.   So you can see the machine boot,
 get virtual media access, etc.

 O/S is CentOS.

 For smaller systems, RAID 1, and for larger DL380 based systems 0+1

 -D


   
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Thursday, March 27, 2008 8:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question

 How do you get notifications ?
 Is this thru one of the add on packages HP sells for the box ?  Which
 
 One
   
 ?
 Could you be more specific about what you mean by a recovery CD
 and hod do you get console access below multi used to do recovery ??

 What is integrated ILO BIOS Access sounds cool.

 What O/S you usin and what made you pick it ?

 What kind and how many RAIDS are you using. The HP site gave like 8
 different RAID controllers and like 20 CPUs to chose from.  How did
 
 you
   
 chose ?

 Thx for sharing !!!

 Darren Wright wrote:
 
 One of the major reasons we use DL320 / DL380's is the ease of
   
 swapping
   
 drives, and the integrated ILO BIOS level access.We can support
 
 remote
   
 sites with ease.
 
 If a drive dies we get a notification, a new one is sent and a non-
   
 techie can replace it with guidance.No onsite visit.   That is
 
 worth
   
 potentially thousands of dollars.
 
 We also leave a recovery CD there that can be inserted if we need to
   
 rebuild the system remotely.   Never had to, but it's worked in the
 
 lab.
   
 -D

 This message was sent from D2 Technology, INC.



   
 
   
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 This message was sent from D2 Technology, INC.


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Re: [asterisk-users] Had it with Dell Garbage - HP Question

I used to have hundred big HP 9000 boxes running HP-UX 11.0.
Having some open th entrailsls of those big boys and due surgery was a 
damn good feeling. The also came with very very good  HW diagnostics and 
had some place you cold send KERNEL Dumps on  troubled system that was 
often a life saver

shadowym wrote:
 You don't have to build Supermicro stuff yourself if you don't want to.
 Most Supermicro dealers do it for you if you buy all the parts from them.

 It's true that what your doing with Dell/HP is paying for emotional support.
 When it comes to PBX's you not getting any value paying for Dell/HP support.
 If your getting good stuff you should never need their hardware replacement
 warranty either.

 -Original Message-
 From: Jesse Molina [mailto:[EMAIL PROTECTED] 
 Sent: Tuesday, March 25, 2008 9:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question


 If you can't troubleshoot a hardware problem, then you should definitely not
 be thinking about this.  Going with a support-yourself plan is not for
 everyone, especially if you don't have good hands-on hardware ability local
 to where the systems are.

 The cost savings can be significant enough that you don't' need to worry
 about third party support.  Just buy some cold-spare systems.  Out of eight
 servers, you can buy three spares and still have money left over -- assuming
 a three year life span for the systems and related support costs.  This is
 true on 1U and 2U systems -- I'm not sure about larger stuff.

 After all, what are Dell/IBM/HP-Compaq going to do for you, other than
 replace the hardware?  Nothing.  They don't support your custom software and
 configurations, just hardware.  Yank the hard drives, RAID controller,
 install in a spare system, and you're up and running again.  Figure out what
 went wrong on the old system later.  If it's under warranty, get it RMAed at
 your leisure.  If it's not, you've got another two spare systems on the
 shelf, waiting there 24x7 just for you.

 Once again, it's not for everyone.  If you don't feel comfortable with it,
 don't do it!  It works for some businesses, not for others.  It depends on
 who is supporting your servers.  If IBM supports your servers, get IBM
 support.  If you support your servers... then why are you paying them to do
 nothing???  Don't pay for emotional support.



 On Tue, Mar 25, 2008 at 06:55:10PM -0400, Al Baker wrote:
   
 ok - but, who do you call for HW problem ? HP has all levels of 
 warranties all depending on how much $$
 you want to spend. What do you do if you buy and install Supermicro ?
 HP also has 24x7 support center , again not for free, what do you do 
 with Supermicro ???
 I am really interested because I hear a lot of folks putting * on them 
 but I never have worked on them while I have put in bunch of HPs really 
 big boxes..
 Thx for sharing your experience

 Matthew Gibson wrote:
 
 I've had good luck with these guys:

 http://rackmountsetc.com/

 supermicro have never failed me yet.


 On Tue, Mar 25, 2008 at 6:03 PM, Jesse Molina [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:


 If you want barebones where you add your own processor, RAM, hard
 drives, and options, try SuperMicro brand servers.  They are
 thousands of dollars less than the big (fat) names like IBM and
 HP/Compaq, but very good quality.

 I've built several clusters of computers with SuperMicro systems.
  They are great if you want to do barebones, clusters, or other
 special projects.

 It just takes a little more time to do the assembly work.

 Try newegg.com http://newegg.com for some sample pricing.



   

   

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[asterisk-users] Question about PCI Slots for DIGIUMs Boards

I had sent this to Digium Sales and cant get a response from them, I 
don't. know what that means..
So I thought I would ask it hear since i know others have struggled with 
this.

I an considering using *your High Density T*1 cards on a number of 
servers we are considering purchasing. The vendor lists that his system has:

PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 
64-bit/100MHz*

Could you please clarify *WHICH* of the above listed *PCI slots* are 
suitable for use with your *High Density T1 cards*. If there is *more 
than one combination that is possible*, could you please indicate *WHICH 
*which high density T1 cards will work in *WHICH* type of PCI slot.

Thanks for any guidance and help since DIGIUM sales seems to be too busy 
to advise a potential customer

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Re: [asterisk-users] Had it with Dell Garbage - HP Question

How do you get notifications ?
Is this thru one of the add on packages HP sells for the box ?  Which One ?
Could you be more specific about what you mean by a recovery CD
and hod do you get console access below multi used to do recovery ??

What is integrated ILO BIOS Access sounds cool.

What O/S you usin and what made you pick it ?

What kind and how many RAIDS are you using. The HP site gave like 8 
different RAID controllers and like 20 CPUs to chose from.  How did you 
chose ?

Thx for sharing !!!

Darren Wright wrote:
 One of the major reasons we use DL320 / DL380's is the ease of swapping 
 drives, and the integrated ILO BIOS level access.We can support remote 
 sites with ease.   
  
 If a drive dies we get a notification, a new one is sent and a non-techie can 
 replace it with guidance.No onsite visit.   That is worth potentially 
 thousands of dollars. 
  
 We also leave a recovery CD there that can be inserted if we need to rebuild 
 the system remotely.   Never had to, but it's worked in the lab.
  
 -D

 This message was sent from D2 Technology, INC.

   
 

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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards

This will be 4 T1s to a card.So, just so I am not confused.
In my original e-mail I said 

I an considering using *your High Density T*1 cards on a number of 
servers we are considering purchasing. The VENDOR lists that his SYSTEM has:

PCI Express*: two x8 slots*, twoo x8 low profile slots*; *PCI-X: 
64-bit/100MHz*

And you list 3 possible PCI slots for your  DIGIUM Quad T1 Cards

PCI or PCI-X 3.3 Volt Slots

PCI 5.0 Volt Slots

PCI-E (PCI Express) slots
==
Here is the Confusion   - 1)  Can ANY of your QUAD cards go in the PCI Express 
- two x8 slots   - Yes or NO ?

   2  Can ANY of your QUAD cards go in the PCI Express 
- two x8 LOW Profile Slots  - Yes or No ?
  
   3) Can your Quad cards you list as fitting a PCI or 
PCI-X 3.3 Volt Slots  fit in  the PCI-X: 64-bit/100MHz slot 
  OR would I have to ask the vendor Is this PCI-X: 
 64-bit/100MHz  definitely a 3.3 Volt Slot ?
   Yes - It would absolutely definitely fit OR NO - 
It will NOT fit   OR   To answer that I need more info  ?

THX for your help
 


Jared Smith wrote:
 On Thu, 2008-03-27 at 06:48 -0400, Al Baker wrote:
   
 I an considering using *your High Density T*1 cards on a number of 
 servers we are considering purchasing. 
 

 Two-port T1 cards?  Four-port T1 cards?

   
 Could you please clarify *WHICH* of the above listed *PCI slots* are 
 suitable for use with your *High Density T1 cards*. If there is *more 
 than one combination that is possible*, could you please indicate *WHICH 
 *which high density T1 cards will work in *WHICH* type of PCI slot.
 

 PCI or PCI-X 3.3 Volt Slots
 ---
 TE412P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation)
 TE212P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation)

 PCI 5.0 Volt Slots
 ---
 TE407P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation)
 TE207P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation)

 PCI-E (PCI Express) slots
 -
 TE420B (Quad Span T1/E1/J1/PRI with DSP-based Echo Cancellation)
 TE220B (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation)

 There are also a variety of models available without the echo
 cancellation modules onboard, but personally I recommend purchasing the
 echo cancellation modules in most situations.

   

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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards

No actually I gave it a LOT of thought and I and even asked two 
different techs who repair PCs
and both said what the to vendors are saying is not sufficiently clear 
that I would
make a purchasing decision based on what you have in hand from them
But, nice try at the cheap shot.

Steve Totaro wrote:
 On Thu, Mar 27, 2008 at 6:48 AM, Al Baker [EMAIL PROTECTED] wrote:
   
 I had sent this to Digium Sales and cant get a response from them, I
  don't. know what that means..
  So I thought I would ask it hear since i know others have struggled with
  this.

  I an considering using *your High Density T*1 cards on a number of
  servers we are considering purchasing. The vendor lists that his system has:

 PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X:
  64-bit/100MHz*

  Could you please clarify *WHICH* of the above listed *PCI slots* are
  suitable for use with your *High Density T1 cards*. If there is *more
  than one combination that is possible*, could you please indicate *WHICH
  *which high density T1 cards will work in *WHICH* type of PCI slot.

  Thanks for any guidance and help since DIGIUM sales seems to be too busy
  to advise a potential customer

 

 You never qualify High Density.

 Maybe Digium thought you might get tired of waiting and find your own
 answer http://en.wikipedia.org/wiki/Peripheral_Component_Interconnect

 Besides that, look here http://store.digium.com/products.php?category_id=2

 You should be able to figure it out on your own with a little thought.

 Thanks,
 Steve Totaro

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Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards