Re: [asterisk-users] fxotune question

[Tilghman Lesher]

On Wednesday 04 June 2008 22:02:19 John Morey wrote:
> Hello,
>
> I've run fxotune at different times but continue to get what seem to be
> strange numbers in /etc/fxotune.conf.  It ends up with:
>
> 5=7,255,251,251,2,255,255,1,255
> 6=7,255,251,251,2,255,255,1,255
> 7=7,255,251,251,2,255,255,1,255
> 8=9,2,250,253,4,252,0,255,255
> 9=4,0,0,0,0,0,0,0,0
> 10=5,0,0,0,0,0,0,0,0
> 11=0,0,0,0,0,0,0,0,0
> 12=0,0,0,0,0,0,0,0,0
> ports 5-10 have lines hooked up to them.  The first four lines seem strange
> when compaired to what others have posted and what ports 9 and 10 have.
>
> Also if I'm reading things right my echo ratios seem to be very
> high.  Running "fxotune -d -b 5 -w 1004" gives the following:
> Dumping module /dev/zap/5
> echo ratio = 0.1759 (1960.0 / 11145.0)
> Which I read to be over 17%.  This seems crazy.  Am I reading this right?
> Where should I start to look for problems?

You might check to see if the tip and ring are reversed in your wiring.  That
can frequently cause weird echo problems.

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[asterisk-users] fxotune question

[John Morey]

Hello,

I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf.  It ends up with:

5=7,255,251,251,2,255,255,1,255
6=7,255,251,251,2,255,255,1,255
7=7,255,251,251,2,255,255,1,255
8=9,2,250,253,4,252,0,255,255
9=4,0,0,0,0,0,0,0,0
10=5,0,0,0,0,0,0,0,0
11=0,0,0,0,0,0,0,0,0
12=0,0,0,0,0,0,0,0,0
ports 5-10 have lines hooked up to them.  The first four lines seem strange
when compaired to what others have posted and what ports 9 and 10 have.

Also if I'm reading things right my echo ratios seem to be very
high.  Running "fxotune -d -b 5 -w 1004" gives the following:
Dumping module /dev/zap/5
echo ratio = 0.1759 (1960.0 / 11145.0)
Which I read to be over 17%.  This seems crazy.  Am I reading this right?
Where should I start to look for problems?

Thanks for any info,

John Morey
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Re: [asterisk-users] Browser based VoIP client? None of them are very full featured

[Erik Anderson]

On Wed, Jun 4, 2008 at 5:52 PM, Bob G <[EMAIL PROTECTED]> wrote:
> None of them have features like hold, transfer, voice mail, dtmf, conference
> as far as I know none of them has caller ID
>
> Only 1ezphone.com has all that and the buttons are programmable for CRM
> features.

Hrm:

- no apparent compatibility with any service other than that which is
offered via 1ezphone
- Frequent spammy emails.
- Dubious claims on website: "...we are going to make the only phone
portal you will every want."
- Some poor person's info revealed on the "User Account" page:
http://1ezphone.com/profile.html
- Revelation of someone's call history: http://1ezphone.com/callhistory.html#

I, for one, won't be giving this a try any time soon.

-Erik

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Re: [asterisk-users] Trouble with Polycom phones

[Kevin Smith]

JR Richardson wrote:
> You mentioned this started happening 3 months ago, what happened then?
>  Network changes, equipment changes, traffic increased, new users
> (downloading allot during the day, surfing porn), wireless
> interference?
>
>   
The initial problem started when our DS3 was throwing errors. Once that 
was resolved, it was fine until about a week later when the problems 
started again...but this time no errors from showing on the DS3.

Otherwise, I will try some other suggestions the next time I am back in 
that office.

Thanks again,
Kevin

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Re: [asterisk-users] 911 via MAX TNT ??

[Andrew Kohlsmith (lists)]

On June 4, 2008 06:20:57 pm Joe Carroll wrote:
> Interestingly enough, on the syslog messages from the TNT we are seeing
> "Called = 911, Q850 Cause = 28, SIP Response = 484"

That really looks like the switch that the TNT is talking to is rejecting the 
number, not the TNT...

-A.

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[asterisk-users] Codec troubles

[Joseph L. Casale]

I have my SIP provider and Astra 480i's set to ulaw, but unless my
Snom M3's aren't set to alaw they sound very bad as they pop and drop out?
Why is this?

Thanks!
jlc
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Re: [asterisk-users] http://1ezphone.com/download = sorry no "s"

[Patrick]

On Wed, 2008-06-04 at 18:01 -0500, Bob G wrote:
> sorry its http://1ezphone.com/download

Anyone ran wireshark on the box running this app? Who's to say this
binary swf is to be trusted? Is the source available somewhere?

Cheers,
Patrick



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Re: [asterisk-users] Avaya IP Phones with *

[Bob G]

We use Polycom 650s on our asterisk and Avaya G700s along with Avaya IP
phones .The sound is good on all of them.

  - Original Message -
  From: "Mark Best"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: Re: [asterisk-users] Avaya IP Phones with *
  Date: Wed, 4 Jun 2008 15:58:45 -0700

Busy Lamp features? How is the sound quality compared to
Polycom/Cisco/Snom etc? Recommend this kind of phone?

(FYI: Doing phone research – while trying to be ‘backwards’-compatible
with an Avaya IP G450/S8700 system.)



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob G
Sent: Wednesday, June 04, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Avaya IP Phones with *

Yes we do everyday here at Google

- Original Message -
From: "Mark Best"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4 Jun 2008 15:24:16 -0700


Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?

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Re: [asterisk-users] queue delay between calls to agents

[Tariq ..]

you can reduce the 5 seconds to any number you wish.. but from a personal 
experience .. if you put the retry to zero.. nothing will change.. i suggest to 
use "1" as your minimum aiting number
Tarek Sawah

 

> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Date: Wed, 4 
> Jun 2008 21:04:01 +0200> Subject: [asterisk-users] queue delay between calls 
> to agents> > Hi,> > I want to reduce the dead time before the queue is 
> calling the next agent. I > see there 5 seconds delay.> It is possible to 
> reduce this time, or what is Asterisk doing within this > timeframe.> > best 
> regards> Thomas> > ___> -- 
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Re: [asterisk-users] http://1ezphone.com/download = sorry no "s"

[Bob G]

sorry its http://1ezphone.com/download


  - Original Message -
  From: "Bob G"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: Re: [asterisk-users] Browser based VoIP client? -
  http://1ezphone.com/downloads
  Date: Wed, 4 Jun 2008 17:46:08 -0500

  you can download a FREE browser softphone and or clcik to call at
  http://1ezphone.com/downloads Let me know if you have any porblems
  and I can help you

- Original Message -
From: "Hilary Miller"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] Browser based VoIP client?
Date: Wed, 4 Jun 2008 14:42:10 -0400


Something that I can put on our internal company website to
replace
our hardware IP phones.

I see many web 2.0 startups offering browser based clients for
their
own service, but I can't seem to find anything that I can use
with my
own PBX. Do I suck at searching google or has the future not
arrived
yet?

Thanks for reading!
--
Just Hil

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Re: [asterisk-users] Avaya IP Phones with *

[Mark Best]

Busy Lamp features? How is the sound quality compared to
Polycom/Cisco/Snom etc? Recommend this kind of phone? 

(FYI: Doing phone research - while trying to be 'backwards'-compatible
with an Avaya IP G450/S8700 system.)

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob G
Sent: Wednesday, June 04, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Avaya IP Phones with *

 

Yes we do everyday here at Google

- Original Message -
From: "Mark Best" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4 Jun 2008 15:24:16 -0700


Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?

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Re: [asterisk-users] 911 via MAX TNT ??

[Mik Cheez]

Cause 28 indicates "Invalid number format".

Joe Carroll wrote:
> See below, we replaced the area code and prefix of with NPANXX for 
> concerns
> 
> Interestingly enough, on the syslog messages from the TNT we are seeing 
> "Called = 911, Q850 Cause = 28, SIP Response = 484"
> 
> 
> Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518", 
> "CALLERID(number)=NPANXX3551") in new stack
> -- Executing [EMAIL PROTECTED]:2] Dial("SIP/NPANXX7604-08c46518", 
> "SIP/To-TNT/3100911") in new stack
> -- Called To-TNT/3100911
> Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY
> -- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
>   == Everyone is busy/congested at this time (1:0/0/1)
>   == Auto fallthrough, channel 'SIP/NPANXX7604-08c46518' status is 
> 'CHANUNAVAIL'
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518", 
> "CDR(userfield)=") in new stack
>  Extension Changed NXX5557604 new state Idle for Notify User NXX5557555
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
> Sent: Wednesday, June 04, 2008 11:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
> 
> The first place you may want to look is in the SYSLOG of the TNT,
> allowing you to see things such as the ISDN error code along with the
> SIP code.  You can try to catch that on the terminal of the TNT, but it
> may make more sense to pipe your syslogs out to an external box, if you
> aren't doing it already.
> 
> JR's suggestion that it may be a limit of the trunk you're using.
> 
> 
> 
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> 

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Re: [asterisk-users] Browser based VoIP client? None of them are very full featured

[Bob G]

None of them have features like hold, transfer, voice mail, dtmf,
conferenceas far as I know none of them has caller ID Only 1ezphone.com
has all that and the buttons are programmable for CRM features.

  - Original Message -
  From: "Tim Panton"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: Re: [asterisk-users] Browser based VoIP client?
  Date: Wed, 4 Jun 2008 21:13:00 +0100



  On 4 Jun 2008, at 21:00, Hilary Miller wrote:

  > EdPimentl wrote:
  >> Have you seen these client?
  >> http://www.mozillavoip.com/
  >> http://tringme.com/
  >> http://www.twoiplink.com/
  >>
  
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
  >
  > I was hoping that there was an open, free, full featured sip client
  > with multiple channels and call transferring created in flash or
  java
  > or something besides an extension. I didn't have my hopes up, and
  I'm
  > grateful for your reply! Thanks!

  Don't get too depressed just yet.

  There is actually a sip client in java (based on the NIST SIP code).
  I don't know if it is full featured - I've never used it.
  https://jain-sip-applet-phone.dev.java.net/

  You won't (yet) find a Flash implementation that talks direct to
  your Asterisk because Flash doesn't support UDP (yet) and
  it doesn't include a VoIP protocol (yet).
  So all the Flash softphones out there have to use a Flash-media
  server
  as a protocol translator to get to 'real' VoIP.

  We (www.phonefromhere.com) have a Java IAX applet that can talk
  direct
  to asterisk, but it isn't
  free. (It also doesn't support multi-channel).

  Tim.


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Re: [asterisk-users] Avaya IP Phones with *

[Bob G]

Yes we do everyday here at Google

  - Original Message -
  From: "Mark Best"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: [asterisk-users] Avaya IP Phones with *
  Date: Wed, 4 Jun 2008 15:24:16 -0700


  Does anyone have any experience getting Avaya phones working with
  Asterisk? (I.E. 9650) BLF etc?

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Re: [asterisk-users] Browser based VoIP client?

[Bob G]

You can download a FREE browser softphone and or cliick to call that
supports UDP athttp://1ezphone.com/download It works well with Asterisk I
use it everyday

  - Original Message -
  From: "Hilary Miller"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: Re: [asterisk-users] Browser based VoIP client?
  Date: Wed, 4 Jun 2008 16:30:00 -0400


  On Wed, Jun 4, 2008 at 4:13 PM, Tim Panton wrote:
  > You won't (yet) find a Flash implementation that talks direct to
  > your Asterisk because Flash doesn't support UDP (yet) and
  > it doesn't include a VoIP protocol (yet).
  > So all the Flash softphones out there have to use a Flash-media
  server
  > as a protocol translator to get to 'real' VoIP.

  Oh, that's very interesting. I'll have to stay tuned, then! Thanks!

  --
  Just Hil

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Re: [asterisk-users] Browser based VoIP client? - http://1ezphone.com/downloads

[Bob G]

you can download a FREE browser softphone and or clcik to call at
http://1ezphone.com/downloads Let me know if you have any porblems and I
can help you

  - Original Message -
  From: "Hilary Miller"
  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
  Subject: [asterisk-users] Browser based VoIP client?
  Date: Wed, 4 Jun 2008 14:42:10 -0400


  Something that I can put on our internal company website to replace
  our hardware IP phones.

  I see many web 2.0 startups offering browser based clients for their
  own service, but I can't seem to find anything that I can use with my
  own PBX. Do I suck at searching google or has the future not arrived
  yet?

  Thanks for reading!
  --
  Just Hil

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[asterisk-users] Avaya IP Phones with *

[Mark Best]

Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?

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Re: [asterisk-users] 911 via MAX TNT ??

[Joe Carroll]

See below, we replaced the area code and prefix of with NPANXX for concerns

Interestingly enough, on the syslog messages from the TNT we are seeing "Called 
= 911, Q850 Cause = 28, SIP Response = 484"


Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555
-- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518", 
"CALLERID(number)=NPANXX3551") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/NPANXX7604-08c46518", 
"SIP/To-TNT/3100911") in new stack
-- Called To-TNT/3100911
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY
-- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/NPANXX7604-08c46518' status is 'CHANUNAVAIL'
-- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518", 
"CDR(userfield)=") in new stack
 Extension Changed NXX5557604 new state Idle for Notify User NXX5557555

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
Sent: Wednesday, June 04, 2008 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

The first place you may want to look is in the SYSLOG of the TNT,
allowing you to see things such as the ISDN error code along with the
SIP code.  You can try to catch that on the terminal of the TNT, but it
may make more sense to pipe your syslogs out to an external box, if you
aren't doing it already.

JR's suggestion that it may be a limit of the trunk you're using.



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[asterisk-users] AST-2008-009: (Corrected subject) Remote crash vulnerability in ooh323 channel driver

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2008-009

   ++
   |  Product   | Asterisk-Addons   |
   |+---|
   |  Summary   | Remote crash vulnerability in ooh323 channel  |
   || driver|
   |+---|
   | Nature of Advisory | Remote crash  |
   |+---|
   |   Susceptibility   | Remote unauthenticated sessions   |
   |+---|
   |  Severity  | Major |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | May 29, 2008  |
   |+---|
   |Reported By | Tzafrir Cohen   |
   |+---|
   | Posted On  | June 4, 2008  |
   |+---|
   |  Last Updated On   | June 4, 2008  |
   |+---|
   |  Advisory Contact  | Mark Michelson  |
   |+---|
   |  CVE Name  | CVE-2008-2543 |
   ++

   ++
   | Description | The ooh323 channel driver provided in Asterisk Addons|
   | | used a TCP connection to pass commands internally. The   |
   | | payload of these packets included addresses of memory|
   | | which were to be freed after the command was processed.  |
   | | By sending arbitrary data to the listening TCP socket,   |
   | | one could cause an almost certain crash since the|
   | | command handler would attempt to free invalid memory.|
   | | This problem was made worse by the fact that the |
   | | listening TCP socket was bound to whatever IP address|
   | | was specified by the "bindaddr" option in ooh323.conf|
   ++

   ++
   | Resolution | The TCP connection used by ooh323 has been replaced with  |
   || a pipe. The effect of this change is that data from   |
   || outside the ooh323 process may not be injected.   |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| N/A   |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| N/A   |
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| N/A   |
   |--+-+---|
   | Asterisk Addons  |1.2.x| All versions prior to |
   |  | | 1.2.9 |
   |--+-+---|
   | Asterisk Addons  |1.4.x| All versions prior to |
   |  | | 1.4.7 |
   |--+-+---|
   |Asterisk Business Edition |A.x.x| N/A   |
   |--+-+---|
   |Asterisk Business Edition   

[asterisk-users] Patch for app_asr.c: DTMF instead of goto

[nik600]

Hi to all

if someone of you is interested on it, i've changed the code of app_asr.c

With these patch you can use the ASR application to play DTMF tones,
so you can have your own AGI application that uses the ASR and manages
the DTMF tones without change the dialplan.

EXAMPLE

exten => 003,1,Ringing
exten => 003,2,Wait(3)
exten => 003,3,Answer
exten => 003,4,ASR(t5000c80l4,100,200:pippo,300:pluto,400:paperino)
exten => 003,5,Read(digito||3)
exten => 003,6,SayDigits(${digito})
exten => 003,7,Wait(30)

The old app_asr will send you to the 200,300 or 400 extension.

With the modified app_asr you will hear (and Asterisk can detects, via
AGI or dialplan) 200,300,400 DTMF tones.

You can find more information here.

http://www.kumbe.it/pagine/dettaglio/34/206.html

Bye

-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser

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[asterisk-users] AST-2008-009: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2008-009

   ++
   |  Product   | Asterisk-Addons   |
   |+---|
   |  Summary   | Remote crash vulnerability in ooh323 channel  |
   || driver|
   |+---|
   | Nature of Advisory | Remote crash  |
   |+---|
   |   Susceptibility   | Remote unauthenticated sessions   |
   |+---|
   |  Severity  | Major |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | May 29, 2008  |
   |+---|
   |Reported By | Tzafrir Cohen   |
   |+---|
   | Posted On  | June 4, 2008  |
   |+---|
   |  Last Updated On   | June 4, 2008  |
   |+---|
   |  Advisory Contact  | Mark Michelson  |
   |+---|
   |  CVE Name  | CVE-2008-2543 |
   ++

   ++
   | Description | The ooh323 channel driver provided in Asterisk Addons|
   | | used a TCP connection to pass commands internally. The   |
   | | payload of these packets included addresses of memory|
   | | which were to be freed after the command was processed.  |
   | | By sending arbitrary data to the listening TCP socket,   |
   | | one could cause an almost certain crash since the|
   | | command handler would attempt to free invalid memory.|
   | | This problem was made worse by the fact that the |
   | | listening TCP socket was bound to whatever IP address|
   | | was specified by the "bindaddr" option in ooh323.conf|
   ++

   ++
   | Resolution | The TCP connection used by ooh323 has been replaced with  |
   || a pipe. The effect of this change is that data from   |
   || outside the ooh323 process may not be injected.   |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| N/A   |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| N/A   |
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| N/A   |
   |--+-+---|
   | Asterisk Addons  |1.2.x| All versions prior to |
   |  | | 1.2.9 |
   |--+-+---|
   | Asterisk Addons  |1.4.x| All versions prior to |
   |  | | 1.4.7 |
   |--+-+---|
   |Asterisk Business Edition |A.x.x| N/A   |
   |--+-+---|
   |Asterisk Business Edition   

[asterisk-users] Asterisk-Addons 1.2.9 and 1.4.7 released; Asterisk-Addons 1.6.0-beta4 now available

[Mark Michelson]

The Asterisk development team has released Asterisk-Addons version 1.2.7, 
1.4.9, 
and 1.6.0-beta4 to address a major security vulnerability in the ooh323 channel 
driver. The releases may be downloaded from http://downloads.digium.com/.

AST-2008-009 details a remote crash vulnerability in the ooh323 channel driver:
  * http://downloads.digium.com/pub/security/AST-2008-009.pdf
  * All users of chan_ooh323 in all versions of Asterisk-Addons are affected

In addition to the security vulnerability, there are a number of bug fixes 
included in this release. Changelogs for the various releases may be found at 
the following addresses:

  * Asterisk-Addons 1.2.9: 
http://svn.digium.com/view/asterisk-addons/tags/1.2.9/ChangeLog?view=markup

  * Asterisk-Addons 1.4.7:
http://svn.digium.com/view/asterisk-addons/tags/1.4.7/ChangeLog?view=markup

  * Asterisk-Addons 1.6.0-beta4:
http://svn.digium.com/view/asterisk-addons/tags/1.6.0-beta4/ChangeLog?view=markup



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Re: [asterisk-users] Lumenvox - Gentoo

[David Backeberg]

Make sure you enable all the USE flags, and then perhaps try
emerge boost
again

I've had times where leaving out a badly named USE flag meant that
critical things didn't end up getting built. A particularly egregious
"must enable all USE flags" case is if you try
emerge ejabberd

Without all the USE flags, especially mod_irc (WTF!) you end up with a
useless daemon. Why would you let anybody emerge a chat daemon with no
support for chat?

On Wed, Jun 4, 2008 at 3:34 PM, Kris Edwards <[EMAIL PROTECTED]> wrote:
> Is anyone running Lumenvox on Gentoo?  My asterisk install has been running
> like a champ for a few years now and I really hate the thoughts of changing
> distros just for Lumenvox.
>
> Here is my issue:
>
> The engine needs the libs from boost.  I emerged boost and noticed that
> there were four libs that the engine were looking for that were not
> installed via portage.
>
> libboost_regex.so.2
> libboost_thread.so.2
> libboost_filesystem.so.2
> libboost_date_time.so.2
>
> Instead, I had the above libs without the .2 at the end.  I created symlinks
> in the engines lib folder.
>
> Now, when I try to execute the bin I get:
>
>
> ./LVSRE_SERVER: symbol lookup error:
> /opt/lumenvox/engine/lib/liblv_lvspeechserver.so: undefined symbol:
> _ZN5boost10filesystem8no_checkERKSs
>
> I am using the redhat package.  I haven't tried rpath or debian yet (which
> I'm about to do now).  Just thought maybe someone might have a thought on
> what I should try.
>
> FYI:  I also tried un-emerging boost and building directly from the official
> release (1.35 I belive).  Perhaps there is a ./configure option I need to
> get this to work right.  I have little experience with redhat and 0
> experience with rpath or debian.  I simply used rpm2tar and moved things
> appropriately.
>
> Thanks!
> -Kris
>
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[asterisk-users] Stuck channels and soft hang up

[Tariq ..]

Greetings.. i'm facing a slight problem i hope.. the management in my call 
center requires using the chanspy 555 to monitor newly hired agents.. and there 
seems a problem where the monitoring extension gets stuck and can't soft hang 
upit .. anyone got a solution for that? it just gets stuck
_
Now you can invite friends from Facebook and other groups to join you on 
Windows Live™ Messenger. Add now.
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Re: [asterisk-users] Browser based VoIP client?

[Hilary Miller]

On Wed, Jun 4, 2008 at 4:13 PM, Tim Panton <[EMAIL PROTECTED]> wrote:
> You won't (yet) find a Flash implementation that talks direct to
> your Asterisk because Flash doesn't support UDP (yet) and
> it doesn't include a VoIP protocol (yet).
> So all the Flash softphones out there have to use a Flash-media server
> as a protocol translator to get to 'real' VoIP.

Oh, that's very interesting. I'll have to stay tuned, then! Thanks!

-- 
Just Hil

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Re: [asterisk-users] Browser based VoIP client?

[Tim Panton]

On 4 Jun 2008, at 21:00, Hilary Miller wrote:

> EdPimentl <[EMAIL PROTECTED]> wrote:
>> Have you seen these client?
>> http://www.mozillavoip.com/
>> http://tringme.com/
>> http://www.twoiplink.com/
>> http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
>
> I was hoping that there was an open, free, full featured sip client
> with multiple channels and call transferring created in flash or java
> or something besides an extension. I didn't have my hopes up, and I'm
> grateful for your reply! Thanks!

Don't get too depressed just yet.

There is actually a sip client in java (based on the NIST SIP code).
I don't know if it is full featured - I've never used it.
https://jain-sip-applet-phone.dev.java.net/

You won't (yet) find a Flash implementation that talks direct to
your Asterisk because Flash doesn't support UDP (yet) and
it doesn't include a VoIP protocol (yet).
So all the Flash softphones out there have to use a Flash-media server
as a protocol translator to get to 'real' VoIP.

We (www.phonefromhere.com) have a Java IAX applet that can talk direct  
to asterisk, but it isn't
free. (It also doesn't support multi-channel).

Tim.


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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Brent Davidson]

Just an update.  I tried updating to the newest Rhino Release firmware 
1.15 and newest stable driver version 2.2.6.  It works OK with 
zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against 
zaptel 1.4.10.1 Asterisk does not see any zap channels.  I'm currently 
running one branch office with the upgraded firmware, driver, 
zaptel-1.4.9.2 and Asterisk-1.4.20.1.  I'll see how everything goes 
there and may upgrade the other offices if it works OK.

Thanks,
Brent

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Re: [asterisk-users] G.722?

[Thomas Kenyon]

Steve Underwood wrote:
> Michael Graves wrote:
>> Which flavor of G.722 has been implemented in Asterisk? And starting
>> with what release version?
>>   
> The only flavour with a defined RTP format is the full 64kbps one.
> 
> Steve
> 
I was going to say strawberry, but to try to answer his other question, 
there is a codec as from release 1.6.0 (coming soon, if you look at the 
changelog there has been a lot of work on it reasonably recently), The 
latest 1.4.x release doesn't have a codec or format interpreter, It is 
listed as a codec, with no translation paths. It will probably work in 
passthrough mode in much the same way as H.264 video. (although I don't 
know this for sure).

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Re: [asterisk-users] Browser based VoIP client?

[Hilary Miller]

EdPimentl <[EMAIL PROTECTED]> wrote:
> Have you seen these client?
> http://www.mozillavoip.com/
> http://tringme.com/
> http://www.twoiplink.com/
> http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox

I was hoping that there was an open, free, full featured sip client
with multiple channels and call transferring created in flash or java
or something besides an extension. I didn't have my hopes up, and I'm
grateful for your reply! Thanks!

-- 
Just Hil

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[asterisk-users] Lumenvox - Gentoo

[Kris Edwards]

Is anyone running Lumenvox on Gentoo?  My asterisk install has been running
like a champ for a few years now and I really hate the thoughts of changing
distros just for Lumenvox.

Here is my issue:

The engine needs the libs from boost.  I emerged boost and noticed that
there were four libs that the engine were looking for that were not
installed via portage.

libboost_regex.so.2
libboost_thread.so.2
libboost_filesystem.so.2
libboost_date_time.so.2

Instead, I had the above libs without the .2 at the end.  I created symlinks
in the engines lib folder.

Now, when I try to execute the bin I get:


./LVSRE_SERVER: symbol lookup error:
/opt/lumenvox/engine/lib/liblv_lvspeechserver.so: undefined symbol:
_ZN5boost10filesystem8no_checkERKSs

I am using the redhat package.  I haven't tried rpath or debian yet (which
I'm about to do now).  Just thought maybe someone might have a thought on
what I should try.

FYI:  I also tried un-emerging boost and building directly from the official
release (1.35 I belive).  Perhaps there is a ./configure option I need to
get this to work right.  I have little experience with redhat and 0
experience with rpath or debian.  I simply used rpm2tar and moved things
appropriately.

Thanks!

-Kris
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Re: [asterisk-users] Browser based VoIP client?

[EdPimentl]

Have you seen these client?
http://www.mozillavoip.com/
http://tringme.com/
http://www.twoiplink.com/
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
(Dated, since project changed names)

BTW, you can also trying to roll your own using old OpenWengo... code.
that was what  did.

-E

On Wed, Jun 4, 2008 at 2:42 PM, Hilary Miller <[EMAIL PROTECTED]> wrote:

> Something that I can put on our internal company website to replace
> our hardware IP phones.
>
> I see many web 2.0 startups offering browser based clients for their
> own service, but I can't seem to find anything that I can use with my
> own PBX. Do I suck at searching google or has the future not arrived
> yet?
>
> Thanks for reading!
> --
> Just Hil
>
>
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[asterisk-users] queue delay between calls to agents

[Thomas Winter]

Hi,

I want to reduce the dead time before the queue is calling the next agent. I 
see there 5 seconds delay.
It is possible to reduce this time, or what is Asterisk doing within this 
timeframe.

best regards
Thomas

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[asterisk-users] disable "send reply" in asterisk voicemail

[Damon Estep]

 

 

I have a need to disable the "Send reply" feature in asterisk voicemail
(1.2) because we have an environment where multiple servers use the same
real-time database for voicemail but the voicemail files are stored on
the individual server that the user registers with. When a user on
server A replies to a VM left by a user on server B the voicemail
message is recorded on server A in our environment.

 

Is there a way to disable the feature in 1.2?

 

I see the configs directive for sendvoicemail=yes|no but see no evidence
that this has impact on the send reply option.

 

Any insight?

 

Thanks!

 

Damon

 

 

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[asterisk-users] Browser based VoIP client?

[Hilary Miller]

Something that I can put on our internal company website to replace
our hardware IP phones.

I see many web 2.0 startups offering browser based clients for their
own service, but I can't seem to find anything that I can use with my
own PBX. Do I suck at searching google or has the future not arrived
yet?

Thanks for reading!
-- 
Just Hil

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Brent Davidson]

Matt Watson wrote:
> Have you tuned rxgain & txgain in Zapata.conf?   
> http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
>  
>
> Also, have you used fxotune to tune each FXO interface?
>
> I believe echo cancellation happens at the Zaptel / DAHDI level, so using 
> Asterisk 1.6 probably isn't going to give you any benefit.
>
>
> --
> Matt Watson
> http://www.mattgwatson.ca
>
>   
FXOTune is apparently not compatible with the R4FXO cards.  Here is the 
output:

fxotune -i
Tuning module /dev/zap/1
Unable to set impedance on fd 4
Failure!
Tuning module /dev/zap/2
Unable to set impedance on fd 4
Failure!
/dev/zap/3 absent: No such device or address
/dev/zap/4 absent: No such device or address
/dev/zap/5 absent: No such file or directory
/dev/zap/6 absent: No such file or directory
/dev/zap/7 absent: No such file or directory
.
. {multiple lines of same basic message edited out}
.
/dev/zap/246 absent: No such file or directory
/dev/zap/247 absent: No such file or directory
/dev/zap/248 absent: No such file or directory
/dev/zap/249 absent: No such file or directory
/dev/zap/250 absent: No such file or directory
/dev/zap/251 absent: No such file or directory
/dev/zap/252 absent: No such file or directory
Unable to tune 2 devices, even though those devices are present


Thanks,
Brent

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Brent Davidson]

Tzafrir Cohen wrote:

On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
  

Brent Davidson wrote:

We're currently using Asterisk 1.4.19, Zaptel 1.4.10, 
Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.
  
Why on earth are you running two layers of echo cancellation - hardware 
and software?  To be honest, I think this is asking for trouble - I've 
seen two occasions where having Oslec and hardware echo cancellation has 
caused significant problems with audio quality - the usual symptoms are 
gaps in the conversation as the hardware cancellation eliminates the 
majority of the echo and the software cancellation subsequently 
eliminates parts of the conversation.



If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
OSLEC in this case) will not be used.

  



Is there any indication of this in Zaptel?This is the output of my 
ztcfg -vv:


Zaptel Version: 1.4.10
Echo Canceller: Oslec
Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)

2 channels to configure.

I removed Oslec when I first installed the R4FXO-EC cards, but echo was 
terrible and made the calls unusable.  My gut instinct is telling me 
that my hardware echo cancellation is not working.


Thanks,
Brent
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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Tzafrir Cohen]

On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
> Brent Davidson wrote:
> > We're currently using Asterisk 1.4.19, Zaptel 1.4.10, 
> > Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.
> 
> Why on earth are you running two layers of echo cancellation - hardware 
> and software?  To be honest, I think this is asking for trouble - I've 
> seen two occasions where having Oslec and hardware echo cancellation has 
> caused significant problems with audio quality - the usual symptoms are 
> gaps in the conversation as the hardware cancellation eliminates the 
> majority of the echo and the software cancellation subsequently 
> eliminates parts of the conversation.

If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
OSLEC in this case) will not be used.

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

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Re: [asterisk-users] Trouble with Polycom phones

[Kevin Smith]

Yes, I was using a name instead of an IP address. And if memory 
servesI *think* it is using TCPprefered...but I could be wrong.

Kevin

Mike wrote:
>>> I have been running into a few issues with Asterisk/polycom and I am
>>> running out of ideas. This problem has been ongoing for the last couple
>>> of weeks. I will try to be as detailed as I can, but I might leave out a
>>> few details. Any suggestions would be greatly appreciated.
>>>   
>
>
>   
>>> Now, the phones lose their registration with Asterisk. 
>>>   
>
> Are you using a numeric IP address or a name for the Asterisk server in the
> Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in
> the numerical IP.
>
> Can't explain it, maybe somebody else can.
>
> Mick
>
>
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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Rob Hillis]

Brent Davidson wrote:
> We're currently using Asterisk 1.4.19, Zaptel 1.4.10, 
> Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.

Why on earth are you running two layers of echo cancellation - hardware 
and software?  To be honest, I think this is asking for trouble - I've 
seen two occasions where having Oslec and hardware echo cancellation has 
caused significant problems with audio quality - the usual symptoms are 
gaps in the conversation as the hardware cancellation eliminates the 
majority of the echo and the software cancellation subsequently 
eliminates parts of the conversation.

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Matt Watson]

Have you tuned rxgain & txgain in Zapata.conf?   
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
 

Also, have you used fxotune to tune each FXO interface?

I believe echo cancellation happens at the Zaptel / DAHDI level, so using 
Asterisk 1.6 probably isn't going to give you any benefit.


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


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: Wednesday, June 04, 2008 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6 vs 1.4?

Is there some location that outlines the major differences between
Asterisk version 1.4 and version 1.6?  I've read through change logs and
several technical discussions, but technical details don't really give
me the big picture.  Basically, is 1.6 more stable than 1.4?  Is it more
efficient?  Does it work better with echo cancelers like Oslec?  I'm
currently using Asterisk as a PBX for our branch offices and will soon
be converting our main office.  Our goal is to be able to have 2 analog
lines at each office, calls come in to each PBX and are routed by VOIP
to a receptionist at one of the offices who then routes calls
appropriately.  We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.  All of our
branch offices have 1MBPS DSL connections and are linked to each other
by VPN's running on our Cisco 1720 routers. Our only problem so far is
with intermittent echo on calls.  Most of the calls have a little echo
right at first, but it goes away almost immediately as the echo canceler
trains.  Every now and then, however, we get a call with terrible echo.
I've put in several e-mails to rhino support asking if the hardware echo
canceler needs something I haven't done but didn't get a response.  I
know echo is just something we have deal with when using analog lines,
but I didn't think it would be this big of a problem.  All of our
offices are in rural areas where digital lines are unavailable so that
is not an option.

Given this setup, is there any reason for me to switch to Asterisk 1.6
or should I stick with 1.4?

Thanks,
Brent Davidson

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Steve Davies]

2008/6/4 Brent Davidson <[EMAIL PROTECTED]>:
[snip]
>  We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
> Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.
[snip]

Just a small aside...

You go to the trouble of building/using Oslec, and then use hardware
EC? Very odd. Does Oslec understand about not doing EC if the hardware
is doing EC? I imagine this is brokered by Zaptel, but am not sure.
Perhaps you are doing double-EC and causing breakage?

Just a random thought.
Steve

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[asterisk-users] Asterisk 1.6 vs 1.4?

[Brent Davidson]

Is there some location that outlines the major differences between 
Asterisk version 1.4 and version 1.6?  I've read through change logs and 
several technical discussions, but technical details don't really give 
me the big picture.  Basically, is 1.6 more stable than 1.4?  Is it more 
efficient?  Does it work better with echo cancelers like Oslec?  I'm 
currently using Asterisk as a PBX for our branch offices and will soon 
be converting our main office.  Our goal is to be able to have 2 analog 
lines at each office, calls come in to each PBX and are routed by VOIP 
to a receptionist at one of the offices who then routes calls 
appropriately.  We're currently using Asterisk 1.4.19, Zaptel 1.4.10, 
Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.  All of our 
branch offices have 1MBPS DSL connections and are linked to each other 
by VPN's running on our Cisco 1720 routers. Our only problem so far is 
with intermittent echo on calls.  Most of the calls have a little echo 
right at first, but it goes away almost immediately as the echo canceler 
trains.  Every now and then, however, we get a call with terrible echo.  
I've put in several e-mails to rhino support asking if the hardware echo 
canceler needs something I haven't done but didn't get a response.  I 
know echo is just something we have deal with when using analog lines, 
but I didn't think it would be this big of a problem.  All of our 
offices are in rural areas where digital lines are unavailable so that 
is not an option.

Given this setup, is there any reason for me to switch to Asterisk 1.6 
or should I stick with 1.4?

Thanks,
Brent Davidson

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Re: [asterisk-users] G.722?

[Steve Underwood]

Michael Graves wrote:
> Which flavor of G.722 has been implemented in Asterisk? And starting
> with what release version?
>   
The only flavour with a defined RTP format is the full 64kbps one.

Steve


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Re: [asterisk-users] 911 via MAX TNT ??

[Mik Cheez]

The first place you may want to look is in the SYSLOG of the TNT, 
allowing you to see things such as the ISDN error code along with the 
SIP code.  You can try to catch that on the terminal of the TNT, but it 
may make more sense to pipe your syslogs out to an external box, if you 
aren't doing it already.

JR's suggestion that it may be a limit of the trunk you're using.

Joe Carroll wrote:
> Hi Mik:
> The TNT is at the ip address 172.16.10.230 and the asterisk box is at 
> 172.16.10.240...
> 
> 
> The trunk group is 3100..   so we send 3100911 to the TNT and get the message 
> below..   I couldn't figure it out..any ideas ??
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
> Sent: Tuesday, June 03, 2008 8:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
> 
> Without knowing more about how you have your TNT set up, typically you'd
> configure your outbound T1's to a specific trunkgroup and prepend that
> trunkgroup number to the phonenumber.
> 
> Should it be assumed that 172.16.10.230 is the address of the TNT?
> 
> Mik
> 
> Joe Carroll wrote:
>> Quick question for the folks using MAX TNTs for aggregators..
>>
>>
>>
>> When I send a call out the MAX I get the following
>>
>>
>>
>> -- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
>>
>>
>>
>> Any ideas on how to make 911 appear as a ten digit number to the device
>> so that it will pass the number out to the PSTN ?
>>
>>
>>
>>
>> 
>>
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[asterisk-users] G.722?

[Michael Graves]

Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?

Thanks,

Michael
--
Michael Graves
mgravesmstvp.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] G.722 over ISDN PRI/BRI

[Matthew Fredrickson]

Simon Hyde wrote:
> Hi,
> 
> G.722 is heavily used by Broadcasters worldwide for wideband voice 
> communications over ISDN. I'd like to be able to receive these G.722 over 
> ISDN 
> calls into an Asterisk exchange (with mostly a view to routing the calls to a 
> Voicemail box where material can be recorded). I have been examining source 
> code for the 3 different ISDN Channels in Asterisk and they all seem to be 
> hard-
> codec to aLaw/uLaw G.711. It looks as though chan_capi *might* support 
> bridging 
> of G.722 data from one ISDN port to another, but not routing to any other 
> source/transcoding/passing to voicemail.
> 
> So I guess my question is, am I correct in the belief that all Asterisk's 
> ISDN 
> channels currently don't support anything other than G.711? How easy would it 
> be to extend one of the ISDN channels to support G.722?

Your belief is correct.  Right now, the ISDN channels (at least in 
chan_zap) G.711 is the only voice codec that is supported.  I'm not sure 
what is going to be necessary to get G.722 working there.  If it's as 
simple as changing the bearer capability, the chan_zap work on top of 
that should be fairly easy.

If you have to implement any of the H.* specs to get it working, that 
will be a bit more trouble.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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[asterisk-users] init.d script no longer uses safe_asterisk

[Paul Belanger]

I noticed safe_asterisk is nolonger used from the init.d script (on
ubuntu) for asterisk-1.6.0-beta9.  I'm curious if there is another
init.d script out there, or even the best way to call safe_asterisk.
Or is safe_asterisk nolonger the script of choice for starting,
restart asterisk.

One of the main reason we like it, is the email notification if it crashes.

Thanks,
PB

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[asterisk-users] Asterisk 1.4.20.1 problems

[Christian Victor]

Hi!

I just upgraded my Asterisk server from 1.4.6 to 1.4.21 and now I experience
some strange behaviour.

1) The Asterisk CLI (asterisk -r) stops responding after some minutes. I
cant CTRL-C or exit the CLI anymore and no activity is shown. Just like if
the connection is interrupted.

2) When I dial into the machine via ISDN (Sangoma A102dx) everything works
okay. But when I hang up it takes about 10 seconds until asterisk recognises
the hangup and jumps to the h extension. pri intense debug shows the
"DISCONNECT" message immediately after I hangup the phone.


Did anybody of you encounter similar problems or do you know what the
reason/solution could be?


Thanks a lot
Christian
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Re: [asterisk-users] 911 via MAX TNT ??

[Kristian Kielhofner]

On 6/3/08, Joe Carroll <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Quick question for the folks using MAX TNTs for aggregators..
>
>
>
> When I send a call out the MAX I get the following….
>
>
>
> -- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
>
>
>
> Any ideas on how to make 911 appear as a ten digit number to the device so
> that it will pass the number out to the PSTN ?
>

I've never used a TNT before but what does your dial pattern matching look like?

If you were using Asterisk, your match would probably look like this:

_NXXNXX

"911" would match NXX but not the remaining digits, hence the 484
Address Incomplete.  I bet your TNT is doing something similar

However:
_NXXNXX
_911

Would work just fine (in Asterisk).  You need to figure out how to do
something similar on your TNT.

-- 
Kristian Kielhofner
NOT sent from my iPhone or Blackberry

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Re: [asterisk-users] connecting 2 FXS together

[Marc Charbonneau]

On Wed, Jun 4, 2008 at 9:04 AM, Steven Howes <[EMAIL PROTECTED]> wrote:
> On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
>> can I connect 2 FXS plugs to the same analog phone ?
>
> No. Fire and death.

Unless you use a 2-lines analog phone :)

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Re: [asterisk-users] connecting 2 FXS together

[Steven Howes]

On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
> can I connect 2 FXS plugs to the same analog phone ?

No. Fire and death.

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

[Tilghman Lesher]

On Wednesday 04 June 2008 02:07:13 Al Baker wrote:
> 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)
>
> 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.

Not they -- me.  And I'm working on transactional support.  If this is really
something you're interested in, then I hope you will test the functionality
when it's ready for that stage of development.

-- 
Tilghman

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[asterisk-users] problem configuring queue

[enediel gonzalez]

Thanks for the answer
My case is the following

queues.conf
[6010]
;fullname = techsupport
strategy = rrmemory
timeout = 10
context = ntech
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
member = SIP/6000


extentions.conf
[ntech]
exten => 6010,1,Answer
exten => 6010,2,Queue(techsupport)
exten => 6010,3,Hangup

6000 is a regular extention, and on 6010 I have configured a queue, using 
another extention if I dial 6010 the queue answer me without problem, in this 
case it only rings on 6000 because it's the only member assigned.

but I need something like
[ntech]

exten => 6010,1,Answer
;exten = s,3,Agi(agi://172.20.3.124/noaction)

exten => 6010,2,Queue(techsupport)

exten => 6010,3,Hangup


I commented the line because I'm not sure exactly how to declare it, basically 
I need to ask some questions to the caller, and when it finished then I want to 
send the call the queue, here is my problem
my application in c# has the following function
public override void Service(AGIRequest request, AGIChannel channel)
{
Asterisk.NET.util.Tools T = new Asterisk.NET.util.Tools();  
  

//T.Channel.SetContext("ntech");
//T.Channel.SetExtension("6010");
//T.Channel.Answer();
//T.Channel.Answer();
//T.Channel.SetExtension("6010");
//T.SetContext("ntech");
//T.SetPriority(1);
//T.SetContext("ntech1");
//T.SetExtension("6000");
//T.SetPriority(100);
//System.Threading.Thread.Sleep(10);
}
I tried all combinations without sucess, I wanted just to call the script, and 
return properly to continue the call on the queue

Thanks in advance for any answer
Greetings
Enediel


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Re: [asterisk-users] 911 via MAX TNT

[JR Richardson]

> When I send a call out the MAX I get the following
> 
> -- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
> 
> Any ideas on how to make 911 appear as a ten digit number to the device so
> that it will pass the number out to the PSTN ?

This is not a max tnt problem, the tnt will pass anything you send to it,
911/411/7 digit/10digit/011 international, the question is, does your PSTN
provider accept 911 call on the trunk your passing the call to?

JR


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

[Steve Totaro]

Less than two minutes of googling:
http://www.asterisk.org/doxygen/1.2/AstDebug.html

Please try to figure out your own problem before sending it to a list
with thousands and thousands of recipients.

Thanks,
Steve Totaro

On Wed, Jun 4, 2008 at 8:17 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> I did not see anything wrong with his English, but maybe your
> understanding of the subject is lacking.  It is like talking to my
> mother about routing or SIP.
>
> People here will often not spoon feed you the answers but give you a "clue".
>
> Google results from his "clue" (thread apply all bt)
>
> http://www.google.com/search?q=thread+apply+all+bt&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a
>
> Again, if you want concise step by step answers, in a timely fashion,
> and are not willing to do research, hire a consultant.
>
> Thanks,
> Steve Totaro
>
> On Wed, Jun 4, 2008 at 3:12 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>> 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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>

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

[Steve Totaro]

I did not see anything wrong with his English, but maybe your
understanding of the subject is lacking.  It is like talking to my
mother about routing or SIP.

People here will often not spoon feed you the answers but give you a "clue".

Google results from his "clue" (thread apply all bt)

http://www.google.com/search?q=thread+apply+all+bt&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a

Again, if you want concise step by step answers, in a timely fashion,
and are not willing to do research, hire a consultant.

Thanks,
Steve Totaro

On Wed, Jun 4, 2008 at 3:12 AM, Al Baker <[EMAIL PROTECTED]> wrote:
> 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] handling jabber status

[Philippe Sultan]

Hi Matt,

On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson <[EMAIL PROTECTED]> wrote:
> I'd be interested to know more about the status abilities as well, we've
> tried to test jabberstatus application, but it doesn't seem to function as
> we expect, it should be returning 0,1,2,3,4,5 based on users current status,
> but switching to away doesn't seem to change it from 0 to 2 .. .
>
> this could be an interesting thread :)

JabberStatus is supposed to retrieve the XMPP status of a buddy, and
store it in a diaplan variable. I just tested it on my Asterisk (1.6)
server.

Here is an example of how to use it :

1234 => {
  JabberStatus(asterisk-gmail,[EMAIL PROTECTED],STATUS);
  if (${STATUS}=1) {
NoOp(User is online and active, ring his Gtalk client.);
Dial(Gtalk/asterisk-gmail/[EMAIL PROTECTED]);
  } else {
NoOp(Prefer the SIP phone);
Dial(SIP/1234);
  }
}

Matt, if you're experiencing some problems with this application, for
example on a 1.4 system, do not hesitate to file a bug report.

Cheers,

Philippe

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[asterisk-users] This AEL vs. Dialplan thing ...

[Gordon Henderson]

Reading this thread with interest..

Curiously enough I asked something similar a while back - got a few 
replies and at the time decided to stick to pure dialplan - but my 
application was a general purpose PBX type of thing, and I didn't want to 
use realtime nor an SQL database... ('embedded' system running from RAM), 
So I have a "core" of dialplan, some hand-crafted macros, contexts, etc. 
then everything else is machine generated - I have many PHP programs that 
write the dialplan for me, based on the plain-text config files I keep (I 
don't read anything in /etc/asterisk, I generate bits of it).

Typically, I'll hand-code a new function, then translate it into a config 
file representation, then PHP code to write the dialplan.

This has worked well for me - and is fast enough to use for test systems 
with a few 100 extensions (although typicall it's in the 20-60 range) - 
where I re-write a lot of dialplan, then send an 'extensions reload' to 
the system (1.2.x) No crashes doing this yet!

So maybe I'm creating a hybrid sort of system where PHP is my AEL :)

I'll give you an example:

Create a "call group" via my GUI - which creates an extension and a list 
of other extensions in that group which are then called at the same time, 
but each extension has a do not disturb flag which has to be honoured...

The line in my "extensions" config file looks like:

   222:100,101,102,103,104,109::CG:House Phones:

Which means extension is 222, members are 100, 101, 102,103,104 and 109, 
it's a Call Group called House Phones..

This generates (gulp - glad I didn't have to type this in!)

;   House Phones
exten => 222,1,Noop(Call Group House Phones)
exten => 222,n,Set(dialString=)
exten => 222,n(cg2220),Set(dnd=${DB(100/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg2221)
exten => 222,n,Set(dialString=${dialString}&${ext100})
exten => 222,n(cg2221),Set(dnd=${DB(101/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg)
exten => 222,n,Set(dialString=${dialString}&${ext101})
exten => 222,n(cg),Set(dnd=${DB(102/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg2223)
exten => 222,n,Set(dialString=${dialString}&${ext102})
exten => 222,n(cg2223),Set(dnd=${DB(103/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg2224)
exten => 222,n,Set(dialString=${dialString}&${ext103})
exten => 222,n(cg2224),Set(dnd=${DB(104/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg2225)
exten => 222,n,Set(dialString=${dialString}&${ext104})
exten => 222,n(cg2225),Set(dnd=${DB(109/doNotDisturb)})
exten => 222,n,GotoIf($["${dnd}" != ""]?cg2226)
exten => 222,n,Set(dialString=${dialString}&${ext109})
exten => 222,n(cg2226),Noop(Calling ${dialString:1})

-- so the above stuff is generated by a PHP loop, and the stuff below was 
hand coded by me, but then put into the PHP file for it to write out as 
part of the process.

exten => 222,n,Noop(callGroup - Checking ringTone: Currently set to 
<${ringTone}>)
exten => 222,n,GotoIf($["${ringTone}" = ""]?noEntryRingTone)
exten => 222,n,Noop(callGroup - ringTone of ${ringTone} already set.)
exten => 222,n,Set(_ringTone=${ringTone})
exten => 222,n,SIPAddHeader(Alert-Info: ${ringTone})
exten => 222,n,Goto(doneRingToneCheck)
exten => 222,n(noEntryRingTone),Noop(callGroup - No entry ringTone set - 
checking internal ringTone)
exten => 222,n,GotoIf($["${internalRingTone}" = ""]?doneRingToneCheck)
exten => 222,n,Noop(callGroup - Internal ringTone of ${ringTone} set.)
exten => 222,n,Set(_ringTone=${internalRingTone})
exten => 222,n,SIPAddHeader(Alert-Info: ${ringTone})
exten => 222,n(doneRingToneCheck),Noop(callGroup - end of ringTone check)
exten => 222,n,Dial(${dialString:1},,wton)


Maybe the last part should be in a macro, but it's funny how things 
evolve!

Gordon

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Re: [asterisk-users] handling jabber status

[Philippe Sultan]

Hi Benoit,

> Anyone already did that (changing jabber status/ status message of many
> accounts)
> or know if it's even remotly possible ??

We discussed that during the last XSF devcon in Brussels. Actually
Asterisk (or any other XMPP client) cannot change the Jabber status on
behalf of another Jabber user, even if you connect it as a component
to your XMPP server. This behaviour is forbidden by the XMPP specs.

To be able to do this, you can use OpenFire along with its Asterisk
plugin, or patch your own XMPP server. I had written a patch for
Jabberd2 some time ago, but I'm not aware of anything that would be
applicable to Ejabberd (written in Erlang).

I'm in the process of extending Asterisk's "Hints" dialplan to XMPP
notifications for authorized users. Those notifications would be
carried over XMPP as message stanza of type 'headline'. That will be a
first step toward implementing PEP (Personal Eventing via Pubsub).
Although I understand that this won't answer your specific need, I
thought you might be interested in knowing this.

I can build a private branch to make this code available if you're
interested in testing it.

Philippe

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[asterisk-users] connecting 2 FXS together

[Joao Ferreira gmail]

Hello all,

this might be a crazy question

can I connect 2 FXS plugs to the same analog phone ?

my reason: I'm expecting that, with this setup, the phone could operate
transparently through the redundant FXS if the main FXS would fail... of
if asterisk is stopped on one of the servers...

the idea is that the users would not even realize one of the asterisks
is not working and the call was routed by the 'spare' asterisk...


|| FXS; asterisk1
PHONE --|shunt   |
|| FXS; asterisk2 (spare)


has anyone tried this ?

thanks in advance

Joao



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Re: [asterisk-users] Any reason to *not* use AEL? (Also, MixMonitor q)

[Gavin Henry]

2008/6/4 Tzafrir Cohen <[EMAIL PROTECTED]>:
> On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
>> What about using RealTime LDAP in 1.6? That woudl be much faster than a 
>> RDBMS.
>
> If performance is such a major issue, why not use explicit queries?
>
> realtime has overhead even in extensions/proiorities where it is not used.

Static will always be faster than any Realtime. But as I understand
it, some things should be kept out of it.

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[asterisk-users] busydetect=yes, busycount=5: hangup automtically without reason, why?

[bilal ghayyad]

Hi All;

Why busydetect=yes caused this autmatic hangup happens
without any reson (while responding to entering the
digits in the IVR) I do not know! And what is the
solution I do not know.

I used busydetect=yes and busycount=5 in zapata.conf
to help me in hangup when detect the busy tone, but I
faced the following problem:

While I am calling to the Asterisk PBX, and during the
response for the IVR and enter the digits and the
code, it hangup by it self (whout any reason).

I do not if I have to increase the busycount for
higher values, or it is related to other paramters or
it is related to the
-DBUSYDETECT_COMPARE_TONE_AND_SILENCE option in the
Asterisk Makefile, even if it is related to that
option, really I do not know what shall I do and what
exactly things mean to be able to set something
resonable.

Any help?
Regards
Bilal


  

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Re: [asterisk-users] Any reason to *not* use AEL? (Also, MixMonitor q)

[Tzafrir Cohen]

On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
> What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.

If performance is such a major issue, why not use explicit queries?

realtime has overhead even in extensions/proiorities where it is not used.

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

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Re: [asterisk-users] Any reason to *not* use AEL? (Also, MixMonitor q)

[Gavin Henry]

What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.


2008/6/3 Sherwood McGowan <[EMAIL PROTECTED]>:
> Mindaugas Kezys wrote:
>> Thank you for your opinion.
>>
>> Then my question would follow: how to build human-friendly system which will
>> use GUI and lets user use that system without messing with .conf files?
>>
>> From my experience large and complicate systems can't be effectivelly
>> managed without Realtime and I see no way how to put AEL into DB. Maybe it's
>> possible?
>>
>> We are storing "exact-match" info into DB and all _X., etc stuff we have in
>> extensions.conf. So no speed issues with large systems.
>>
>> Also: Any reason to "not" use extensions.conf?
>>
>> What AEL can do better then extensions.conf?
>>
>> Many people still use vi. Because it can do everything what they want. Same
>> here with extensions.conf.
>>
>> Regards,
>> Mindaugas Kezys
>> http://www.kolmisoft.com
>>
>>
>>
>>> -Original Message-
>>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>>> [EMAIL PROTECTED] On Behalf Of Steve Murphy
>>> Sent: Tuesday, June 03, 2008 9:02 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] Any reason to *not* use AEL? (Also,
>>> MixMonitor q)
>>>
>>> On Tue, 2008-06-03 at 09:33 -0500, Sherwood McGowan wrote:
>>>
 Mindaugas Kezys wrote:

> Does Asterisk Realtime support AEL?
>
>
>
> Regards,
>
> Mindaugas Kezys
>
> http://www.kolmisoft.com
>
>
>
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of
> *Gonzalo Servat
> *Sent:* Tuesday, June 03, 2008 5:07 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Any reason to *not* use AEL? (Also,
> MixMonitor q)
>
>
>
> On Tue, Jun 3, 2008 at 10:41 AM, Eric Wieling <[EMAIL PROTECTED]
> > wrote:
>
> AEL in 1.4 was the first version of AEL that most people
>
>>> consider
>>>
> "stable".  Since not many people uses AEL, you won't get nearly
>
>>> as
>>>
> much
> (if any) community support compared to if you are using the
> non-AEL syntax.
>
>
> Really? Why would anyone want to write a dialplan using the old
> extensions.conf syntax? That sort of syntax personally drove me
>
>>> nuts
>>>
> (and real messy). I've got my entire dialplan on AEL (using
>
>>> Asterisk
>>>
> 1.6.0).
>
>
> -
>
 Not sure what you mean, but if you mean realtime dialplan, then no,
 you can't use AEL for that. However, we might wish to see if Murf
 knows if this can be done.


>>> extensions.conf is like assembler; it's a very strict, line per
>>> instruction format, 4 fields per line, that is able to be read in by
>>> normal config file parsers. It is in turn compiled into the internal
>>> asterisk data structures.
>>>
>>> AEL is more free form. Storing the dial plan in AEL format in a db
>>> would be pretty useless. However, the extensions.conf isn't so bad in a
>>> db, as it still has the 4 columns, row per instruction sort of format.
>>>
>>> But in general, I have to ask, as a programmer, if it's really, really
>>> a good idea to store code in a db. The dialplan is a mixture of both
>>> dialplan code and data, in the form of extensions.
>>>
>>> But storing dialplan "code", as in a sequence of application calls, is
>>> a slow way to execute your dialplan code.
>>>
>>> And storing patterned extensions (extensions starting with _, like
>>> _10X or whatever), is a really slow way to match pattern
>>> extensions. My advise to everyone is this: Realtime is great, but don't
>>> store extension patterns in there, and don't store your dialplan code
>>> in there, if you can help it. It'd be much better to use your db to
>>> store 'exact' extension data. Trying to find the best pattern match via
>>> realtime is excruciatingly slow, as it calls up every extension in the
>>> db for that context, and then decides on the best match.  DB's do a
>>> great job at storing large numbers of uniquely keyed data that you can
>>> find via exact matches. So, use a general exten patten in your
>>> dialplan, and then do a DB() lookup from there.
>>>
>>> If you find a bug in your dialplan code, you've got to change it in two
>>> places, in the realtime db, and you'd best have it in your original
>>> source as well, in case you need to reload/recover your db or whatever.
>>> A DB is a lousy source-code control system. Use cvs or subversion or
>>> git or something to store your dialplan code instead. That way, you can
>>> back out change sets, etc, and track your changes in a much more
>>> practical way.
>>>
>>> Just my two cents.
>>>
>>> murf
>>>
>>> --
>>> Steve Murphy
>>> Software Developer
>>> Digium
>>>
>>
>>
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Re: [asterisk-users] Queue is sending calls to Agents even when they are in use

[Thomas Winter]

On Tuesday 03 June 2008 23:22, Atis Lezdins wrote:

> chan_agent with AgentCallbackLogin was working but not completely
> stable for my dialplan which was quite heavy when I was on 1.2,
> however you may try that out. Or just upgrade to 1.4 (or even 1.6 and
> try state_interface)

Iam using API Action QueueAdd through an WebGUI and quite happy with this.
I hope I can jump from 1.2 to 1.6 without touching 1.4:)

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Re: [asterisk-users] Error Wile starting AsterFax

[Paul Hales]

You really should discuss this at the Asterfax forums:

http://forums.asteriskit.com.au/

later,

PaulH


On Wed, 2008-06-04 at 14:17 +0530, Sukhbir Singh wrote:
> Hi All,
>  
>I am getting following error when i start AsterFax:
>  
>Please help me to solve this issue:
>  
> [EMAIL PROTECTED] asterfax]# ./asterfax.sh
> log4j: Threshold ="null".
> log4j: Retreiving an instance of org.apache.log4j.Logger.
> log4j: Setting [au.com.noojee.asterfax] additivity to [false].
> log4j: Level value for au.com.noojee.asterfax is  [DEBUG].
> log4j: au.com.noojee.asterfax level set to DEBUG
> log4j: Class name: [org.apache.log4j.ConsoleAppender]
> log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
> log4j: Setting property [conversionPattern] to [%d{ABSOLUTE} %p %m%n].
> log4j: Adding appender named [STDOUT] to category
> [au.com.noojee.asterfax].
> log4j: Class name: [org.apache.log4j.DailyRollingFileAppender]
> log4j: Setting property [file] to [logs/asterfax.log].
> log4j: Setting property [append] to [true].
> log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
> log4j: Setting property [conversionPattern] to [%d{DATE} %p %m%n].
> log4j: setFile called: logs/asterfax.log, true
> log4j: setFile ended
> log4j: Appender [FILE] to be rolled at midnight.
> log4j: Adding appender named [FILE] to category
> [au.com.noojee.asterfax].
> log4j: Retreiving an instance of org.apache.log4j.Logger.
> log4j: Setting [au.com.noojee.asterfax.outbound.SubmitFax] additivity
> to [false].
> log4j: Level value for au.com.noojee.asterfax.outbound.SubmitFax is
> [DEBUG].
> log4j: au.com.noojee.asterfax.outbound.SubmitFax level set to DEBUG
> log4j: Adding appender named [STDOUT] to category
> [au.com.noojee.asterfax.outbound.SubmitFax].
> log4j: Adding appender named [FILE] to category
> [au.com.noojee.asterfax.outbound.SubmitFax].
> log4j: Retreiving an instance of org.apache.log4j.Logger.
> log4j: Setting [Audit] additivity to [false].
> log4j: Level value for Audit is  [INFO].
> log4j: Audit level set to INFO
> log4j: Class name: [org.apache.log4j.DailyRollingFileAppender]
> log4j: Setting property [file] to [logs/audit.log].
> log4j: Setting property [append] to [true].
> log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
> log4j: Setting property [conversionPattern] to [%d{DATE}, %m%n].
> log4j: setFile called: logs/audit.log, true
> log4j: setFile ended
> log4j: Appender [AUDIT] to be rolled at midnight.
> log4j: Adding appender named [AUDIT] to category [Audit].
> log4j: Level value for root is  [WARN].
> log4j: root level set to WARN
> log4j: Adding appender named [STDOUT] to category [root].
> log4j: Adding appender named [FILE] to category [root].
> 14:10:42,419 INFO AsterFax version 1.0-beta9 starting.
> 14:10:42,509 INFO Debugging Enabled
> 14:10:42,509 INFO Debug Preserve Temp Files enabled
> 14:10:42,509 INFO Debug TxFax enabled
> 14:10:42,510 INFO Debug Converter enabled
> 14:10:42,512 INFO Loading TX handler
> au.com.noojee.asterfax.handler.DefaultFaxHandler
> 14:10:42,514 INFO Loading RX handler
> au.com.noojee.asterfax.handler.DefaultFaxHandler
> 14:10:42,518 INFO Loaded Channel Zap/g0 Both from-internal priority(1)
> count(1)
> 14:10:42,521 INFO Loaded converter for image/pdf pdf
> 14:10:42,522 INFO Loaded converter for application/pdf pdf
> 14:10:42,523 INFO Loaded converter for application/msword doc;dot
> 14:10:42,523 INFO Loaded converter for application/rtf rtf
> 14:10:42,524 INFO Loaded converter for text/rtf rtf
> 14:10:42,524 INFO Loaded converter for
> application/vnd.oasis.opendocument.text odt;ott
> 14:10:42,525 INFO Loaded converter for application/vnd.sun.xml.writer
> sxw;stw
> 14:10:42,526 INFO Loaded converter for
> application/vnd.stardivision.writer sdw
> 14:10:42,526 INFO Loaded converter for application/msexcel xls
> 14:10:42,527 INFO Loaded converter for image/postscript ps
> 14:10:42,527 INFO Loaded converter for application/postscript
> 14:10:42,530 INFO Loaded converter (built-in) for text/plain
> 14:10:42,531 INFO Loaded converter (built-in) for text/html
> 14:10:42,532 INFO Loaded converter (built-in) for image/tiff
> 14:10:42,532 DEBUG Working Directory: /usr/lib/asterfax
> 14:10:42,532 DEBUG class path: AsterFax.jar
> 14:10:42,746 DEBUG OutBound FaxManager starting.
> 14:10:42,771 ERROR /var/spool/asterfax/tmp (Is a directory)
> java.io.FileNotFoundException: /var/spool/asterfax/tmp (Is a
> directory)
> at java.io.FileInputStream.open(Native Method)
> at java.io.FileInputStream.(FileInputStream.java:106)
> at
> au.com.noojee.asterfax.messagestore.FileMimeMessage.(FileMimeMessage.java:116)
> at
> au.com.noojee.asterfax.outbound.FaxManagerOutBound.load(FaxManagerOutBound.java:242)
> at
> au.com.noojee.asterfax.outbound.FaxManagerOutBound.loadAndstart(FaxManagerOutBound.java:83)
> at
> au.com.noojee.asterfax.AsterFax.startThreads(AsterFax.java:187)
> at au.com.noojee.asterfax.AsterFax

[asterisk-users] Error Wile starting AsterFax

[Sukhbir Singh]

Hi All,

   I am getting following error when i start AsterFax:

   Please help me to solve this issue:

[EMAIL PROTECTED] asterfax]# ./asterfax.sh
log4j: Threshold ="null".
log4j: Retreiving an instance of org.apache.log4j.Logger.
log4j: Setting [au.com.noojee.asterfax] additivity to [false].
log4j: Level value for au.com.noojee.asterfax is  [DEBUG].
log4j: au.com.noojee.asterfax level set to DEBUG
log4j: Class name: [org.apache.log4j.ConsoleAppender]
log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
log4j: Setting property [conversionPattern] to [%d{ABSOLUTE} %p %m%n].
log4j: Adding appender named [STDOUT] to category [au.com.noojee.asterfax].
log4j: Class name: [org.apache.log4j.DailyRollingFileAppender]
log4j: Setting property [file] to [logs/asterfax.log].
log4j: Setting property [append] to [true].
log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
log4j: Setting property [conversionPattern] to [%d{DATE} %p %m%n].
log4j: setFile called: logs/asterfax.log, true
log4j: setFile ended
log4j: Appender [FILE] to be rolled at midnight.
log4j: Adding appender named [FILE] to category [au.com.noojee.asterfax].
log4j: Retreiving an instance of org.apache.log4j.Logger.
log4j: Setting [au.com.noojee.asterfax.outbound.SubmitFax] additivity to 
[false].
log4j: Level value for au.com.noojee.asterfax.outbound.SubmitFax is  [DEBUG].
log4j: au.com.noojee.asterfax.outbound.SubmitFax level set to DEBUG
log4j: Adding appender named [STDOUT] to category 
[au.com.noojee.asterfax.outbound.SubmitFax].
log4j: Adding appender named [FILE] to category 
[au.com.noojee.asterfax.outbound.SubmitFax].
log4j: Retreiving an instance of org.apache.log4j.Logger.
log4j: Setting [Audit] additivity to [false].
log4j: Level value for Audit is  [INFO].
log4j: Audit level set to INFO
log4j: Class name: [org.apache.log4j.DailyRollingFileAppender]
log4j: Setting property [file] to [logs/audit.log].
log4j: Setting property [append] to [true].
log4j: Parsing layout of class: "org.apache.log4j.PatternLayout"
log4j: Setting property [conversionPattern] to [%d{DATE}, %m%n].
log4j: setFile called: logs/audit.log, true
log4j: setFile ended
log4j: Appender [AUDIT] to be rolled at midnight.
log4j: Adding appender named [AUDIT] to category [Audit].
log4j: Level value for root is  [WARN].
log4j: root level set to WARN
log4j: Adding appender named [STDOUT] to category [root].
log4j: Adding appender named [FILE] to category [root].
14:10:42,419 INFO AsterFax version 1.0-beta9 starting.
14:10:42,509 INFO Debugging Enabled
14:10:42,509 INFO Debug Preserve Temp Files enabled
14:10:42,509 INFO Debug TxFax enabled
14:10:42,510 INFO Debug Converter enabled
14:10:42,512 INFO Loading TX handler 
au.com.noojee.asterfax.handler.DefaultFaxHandler
14:10:42,514 INFO Loading RX handler 
au.com.noojee.asterfax.handler.DefaultFaxHandler
14:10:42,518 INFO Loaded Channel Zap/g0 Both from-internal priority(1) count(1)
14:10:42,521 INFO Loaded converter for image/pdf pdf
14:10:42,522 INFO Loaded converter for application/pdf pdf
14:10:42,523 INFO Loaded converter for application/msword doc;dot
14:10:42,523 INFO Loaded converter for application/rtf rtf
14:10:42,524 INFO Loaded converter for text/rtf rtf
14:10:42,524 INFO Loaded converter for application/vnd.oasis.opendocument.text 
odt;ott
14:10:42,525 INFO Loaded converter for application/vnd.sun.xml.writer sxw;stw
14:10:42,526 INFO Loaded converter for application/vnd.stardivision.writer sdw
14:10:42,526 INFO Loaded converter for application/msexcel xls
14:10:42,527 INFO Loaded converter for image/postscript ps
14:10:42,527 INFO Loaded converter for application/postscript
14:10:42,530 INFO Loaded converter (built-in) for text/plain
14:10:42,531 INFO Loaded converter (built-in) for text/html
14:10:42,532 INFO Loaded converter (built-in) for image/tiff
14:10:42,532 DEBUG Working Directory: /usr/lib/asterfax
14:10:42,532 DEBUG class path: AsterFax.jar
14:10:42,746 DEBUG OutBound FaxManager starting.
14:10:42,771 ERROR /var/spool/asterfax/tmp (Is a directory)
java.io.FileNotFoundException: /var/spool/asterfax/tmp (Is a directory)
at java.io.FileInputStream.open(Native Method)
at java.io.FileInputStream.(FileInputStream.java:106)
at 
au.com.noojee.asterfax.messagestore.FileMimeMessage.(FileMimeMessage.java:116)
at 
au.com.noojee.asterfax.outbound.FaxManagerOutBound.load(FaxManagerOutBound.java:242)
at 
au.com.noojee.asterfax.outbound.FaxManagerOutBound.loadAndstart(FaxManagerOutBound.java:83)
at au.com.noojee.asterfax.AsterFax.startThreads(AsterFax.java:187)
at au.com.noojee.asterfax.AsterFax.(AsterFax.java:114)
at au.com.noojee.asterfax.AsterFax.main(AsterFax.java:297)
14:10:42,772 INFO Shutting down AsterFax due to IOException.
14:10:42,773 INFO AsterFax has shutdown.

Thanks in Advance,
Sukhbir





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

[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 just stops working...

[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] Mysql and extensions.conf

[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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