[asterisk-users] [OT]I like this community

[Rony Ron]

Hi @ all,
i like this community,
i don't think that there is any place on this planet from where emails 
are not coming directed to this community,
if governments were profiting to each other like the members of this 
community do,
there would be no poor on this planet,
there would be no war on this planet,
there would no deseases on this planet,
Thanks to everybody,
warm regards,
2R

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Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5

[Rony Ron]

Great !
thank you very much for your job!
BR,

Matt Florell a écrit :
> Hello,
>
> We've released another update to our VICIDIAL/astGUIclient call center
> suite: 2.0.5
>
> http://astguiclient.sf.net/
>
> The call center suite client applications run on most modern web
> browsers on almost any GUI-capable operating system, and it includes
> the VICIDIAL call center suite.
> This package is free and AGPLv2.
> This package is geared towards Asterisk installations with SIP,IAX or
> Zap phones and Zaptel, IAX or SIP trunks.
>
> For this release, we have added hundreds of new features including
> Asterisk phone, trunk and DID configuration through the VICIDIAL web
> interface. We have also tested the suite on Asterisk versions through
> 1.2.30.2 and 1.4.21.2.
>
> All client web-apps and administration pages are available in English,
> Spanish, Greek, German, Italian and French, with rough translations of
> Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch
> for the client web-apps only.
>
> Check out the project blog for more information:
> http://astguiclient.blogspot.com
>
> Let me know what you think.
>
> Thanks,
>
>
> MATT---
>
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Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY

[Rony Ron]

Hey !
this can drive to heart attacks 

randulo a écrit :
> Nice one, Olle ! :)
>
> On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson  wrote:
>   
>> * NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
>> VIDEO TO MICROBLOGGING!
>> 
>
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Re: [asterisk-users] Dahdi caused Kernel to segfault

[Rony Ron]

Hi
the same happened here also with different distros (ubuntu and fedora 9)
each time i run "dahdi start" the kernel crash.
i was  using the dahdi from trunk
regards,

David fire a écrit :
> do you have any dahdi card ???
> if not edit /etc/dahdi/modules so it dosent load any modules.
> David
>
> 2009/1/25 broadband Voice  >
>
> More information
>  
> service dahdi start
> Loading DAHDI hardware modules:
> FATAL: Module dahdi not found.
>   wct4xxp: [  OK  ]
>
> Error: missing /dev/dahdi!
> [r...@newmark1 ~]#
>  
>
> On Sun, Jan 25, 2009 at 3:31 PM, broadband Voice
> mailto:broadbandvo...@gmail.com>> wrote:
>
> I had several panic attacks after upgrading to 1.4.22 but now
> we have no dial tone on the T1. Urgent production system.
>
> On Tue, Jan 13, 2009 at 5:13 AM, Benoit
> mailto:maver...@maverick.eu.org>>
> wrote:
>
>
> Personnaly, i had recently encountered  a global machine
> check exception
> with
> two cards (TE220p and B410) and many kernel panic with
> mISDN (mostly if
> i tried to unload it).
>
> Dahdi still hasn't failed me (directly)
>
> Thomas Kenyon a écrit :
> > Yesterday, a low-duty production server that I maintain
> core-dumped. At
> > the time there were only around 2 calls going through it.
> >
> > The strace on the screen made it look like it was caused
> by Dahdi.
> >
> > The machine is running
> >
> > asterisk-1.6.0.3
> > dahdi-linux-2.1.0.3
> > dahdi-tools-2.1.0.2
> > asterisk-addons-1.6.0
> >
> > Kernel version 2.6.28
> >
> > There is a genuine TDM400P (populated witrh 2xFXO cards
> and 2xFXS cards.
> >
> > Has anyone had a similar issue?
> >
> > This has only happened once, but I am a bit worried.
> >
> >
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> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
> 
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Re: [asterisk-users] disable auth between two asterisk

[Rony Ron]

Hello list,
i wanted to setup a small asterisk+ss7 lab this weekend and just installed
asterisk-trunk+ dahdi-complete+libss7+libpri
i had only a sangoma A101 card so i used it and 48h after i'm still
unable to make the card work in that config.

i tried to patch the sangoma drivers thinking that it was just a
matter of find_replace(zaptel BY dahdi) but i discovered that i should
also do a find_replace(zt_ BY dahdi_) and also
find_replace(INSTALL_DIR/kernel BY
INSTALL_DIR/kernel/include/drivers/dahdi) after all that the sangoma
drivers still fails to install due to some declaration that are in the
zaptel.h and not in the file  /include/drivers/dahdi/user.h ... THE
END of tries i give up and go to bed !

Anyone know about any patch of sangoma drivers that support directly dahdi ?
regards,

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Re: [asterisk-users] Beginner Questions part II

[Rony Ron]

Hi John,
*for the first part:
you can create 3 contexts: internal,external and main
in your internal context you put your internal extension
in the external context you send the send the XXX-XXX- to the 
providers trunk
and in the main context you just include the internal context (first) 
then the external context.

** second part:
check here: 
http://www.jackenhack.com/blog/archives/2005/09/26/adding-blacklist-to-an-asteriskhome-pbx-voip-server/

*BR,*

John Koenig a écrit :
> I should start with a thank you to the list for helping me getting up 
> and running with Asterisk about a week ago.  I have been happily 
> fiddling with Asterisk since then :).
>
> I am working on adding a couple features to my dialplan.  My setup 
> involves my asterisk box connecting to another third party sip 
> provider.  I configured the extra trunk and there are no issues passing 
> calls through their systems.  As it stands right now, I setup a calling 
> rule that matches the pattern 9-XXX-XXX-, stripes off the 9, and 
> then passes the call through to the third party.  I would like to just 
> dial XXX-XXX- without having to dial the extra 9.  Is there a way 
> that I can configure asterisk so that I check to see if the extension 
> exists on my box first, if it does then pickup and if it doesn't then 
> forward the call onto the third party?  If so, how?
>
> The other feature I am looking to add is *67 caller id blocking.  Am I 
> right in thinking that I first would configure an incoming call rule 
> that matches to *67- and then pass unknown as the caller id 
> from there?
>
> Any help is greatly appreciated.  Even if it is just pointing me in the 
> right direction in regards to reading material.
>
> Thanks,
>
> John
>
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Re: [asterisk-users] Asterisk VOIP Jobs version 2 Launched!

[Rony Ron]

Hello all,
please, is it possible to which party has hangup a call?
if yes, please tell me how ?
thanks,


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Re: [asterisk-users] Toll fraud detection/password script

[Rony Ron]

Thanks for your contrib

On Nov 22, 2007 2:56 PM, J. Oquendo <[EMAIL PROTECTED]> wrote:
>
> So I was bored yesterday and tried solving a few
> problems with one stone:
>
> 1) Notify me of potential brute forcers (multiple attempts
> to register multiple numbers from one address)
> 2) Notify me of (l)users who are having password issues
>
> So I whipped up a simple script to run in cron and
> notify me that UserX from X_IP_Space had X amout of
> password issues. I'm currently running this from
> cron and it works fine. My personal version is
> modified to block (l)users after 10 failures on
> 2 separate accounts or 50 failures on one account.
>
> Methodology is, if someone hasn't complained within
> two minutes of something happening to their phones
> that they can't log in, then they won't need to
> use that phone right now. Let them call in and
> complain...
>
> http://www.infiltrated.net/scripts/astrap
>
> --
> =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
> J. Oquendo
> SGFA #579 (FW+VPN v4.1)
> SGFE #574 (FW+VPN v4.1)
>
> echo c2lsQGluZmlsdHJhdGVkLm5ldAo=|\
> python -c "import sys; print sys.stdin.read().decode('base64')"
>
> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xF684C42E
>
>
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Re: [asterisk-users] PRI span configuration - span remains down

[Rony Ron]

gt; <  Ext: 1  Cause: Requested channel not available
> > (44), class = Network Congestion (resource unavailable) (2) ]
> > <  Cause data 1: 18 (24)
> > -- Processing IE 8 (cs0, Cause)
> > q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> > Sending Receiver Ready (4)
> > voip1*CLI>
> >> [ 02 01 01 08 ]
> > voip1*CLI>
> >> Supervisory frame:
> >> SAPI: 00  C/R: 1 EA: 0
> >>  TEI: 000EA: 1
> >> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> >> N(R): 004 P/F: 0
> >> 0 bytes of data
> > -- Restarting T203 counter
> > -- Restarting T203 counter
> > -- Channel 0/6, span 1 got hangup, cause 44
> > -- Forcing restart of channel 0/6 on span 1 since channel reported
> in use
> > voip1*CLI>
> >> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
> > voip1*CLI>
> >> Informational frame:
> >> SAPI: 00  C/R: 0 EA: 0
> >>  TEI: 000EA: 1
> >> N(S): 008   0: 0
> >> N(R): 004   P: 0
> >> 13 bytes of data
> > -- Restarting T203 counter
> > Stopping T_203 timer
> > Starting T_200 timer
> >> Protocol Discriminator: Q.931 (8)  len=13
> >> Call Ref: len= 2 (reference 0/0x0) (Originator)
> >> Message type: RESTART (70)
> >> [18 03 a9 83 86]
> >> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare:
> 0  Exclusive  Dchan: 0
> >>ChanSel: Reserved
> >>   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> >>   Ext: 1  Channel: 6 ]
> >> [79 01 80]
> >> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
> Channel (0) ]
> > voip1*CLI>
> > < [ 00 01 01 12 ]
> > voip1*CLI>
> > < Supervisory frame:
> > < SAPI: 00  C/R: 0 EA: 0
> > <  TEI: 000EA: 1
> > < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> > < N(R): 009 P/F: 0
> > < 0 bytes of data
> > -- ACKing all packets from 7 to (but not including) 9
> > -- ACKing packet 8, new txqueue is -1 (-1 means empty)
> > -- Since there was nothing left, stopping T200 counter
> > -- Nothing left, starting T203 counter
> > -- Restarting T203 counter
> > NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
> > NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
> > -- Hungup 'Zap/6-1'
> > [Oct 25 18:01:46] NOTICE[20956]: cdr.c:434 ast_cdr_free: CDR on
> > channel 'Zap/6-1' not posted
> >   == Everyone is busy/congested at this time (1:0/0/1)
> > -- Executing [EMAIL PROTECTED]:7]
> > ResetCDR("SIP/charlie59-082bc890", "w") in new stack
> > -- Executing [EMAIL PROTECTED]:8]
> > NoCDR("SIP/charlie59-082bc890", "") in new stack
> > -- Executing [EMAIL PROTECTED]:9]
> > Answer("SIP/charlie59-082bc890", "") in new stack
> > -- Executing [EMAIL PROTECTED]:10]
> > PlayTones("SIP/charlie59-082bc890", "congestion") in new stack
> >   == Auto fallthrough, channel 'SIP/charlie59-082bc890' status is
> 'CHANUNAVAIL'
> > -- Executing [EMAIL PROTECTED]:1]
> > Hangup("SIP/charlie59-082bc890", "") in new stack
> >   == Spawn extension (route-ext-ycmcr, h, 1) exited non-zero on
> > 'SIP/charlie59-082bc890'
> >
> > As I say, I've asked a separate question on this, so I don't really
> > want to end up with two thread on the one problem :)
> >
> > Thanks
> >
> > Dave
> >
> > On 10/25/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
> >> Rony Ron wrote:
> >>> Hello,
> >>> Quoting Digium Support:
> >>> "The TE110P has been discontinued and replaced in our product lineup
> with
> >>> the TE120P, which features many overall improvements and does not
> suffer
> >>> from the HDLC Abort/Bad FCS problems that the TE110P did."
> >> Although this is true ( :-) ) I think that it is likely his problem is
> >> not related to this.  Can you post a "pri intense debug span x" for the
> >> span in question?
> >>
> >> Matthew Fredrickson
> >>
> >>> On 10/25/07, David Kennedy <[EMAIL PROTECTED]> wrote:
> >>

Re: [asterisk-users] PRI span configuration - span remains down

[Rony Ron]

Hello,
Quoting Digium Support:
"The TE110P has been discontinued and replaced in our product lineup with
the TE120P, which features many overall improvements and does not suffer
from the HDLC Abort/Bad FCS problems that the TE110P did."

Better switch to TE120P,

On 10/25/07, David Kennedy <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I'm trying to connect to Telewest/Virgin Media with a TE110P using
> asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
> appears as
>
> PRI span 1/0: Provisioned, Down, Active
>
> My zapata.conf is currently
> ---
> [channels]
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> switchtype=euroisdn
> contect=from-pri
> signalling=pri_cpe
> group=1
> channel => 1-15
> channel => 17-31
> ---
>
> zaptel.conf is
> ---
> span=1,1,0,ccs,hdb3,crc4
> dchan=16
> bchan=1-15,17-31
> loadzone=uk
> defaultzone=uk
> ---
>
> I'm in London and the server is in Manchester, so I can't look at the
> server directly, but when we first started setting it up, apparently a
> pair of cables were the wrong way round, so the card was in a RED
> alarm state. We've switched the cables and now the card is OK. We did
> have a lot of IRQ misses, so we've upgraded the kernel and now the
> accuracy reported by zttest is about 99.98%. Telewest have checked the
> line for faults and have reported that it's fine, but I just can't get
> it working.
>
> Does anyone have any ideas/suggestions?
>
> Thanks,
>
> Dave
>
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[asterisk-users] Free help

[Rony Ron]

Hello all,
i would like to have references so i'm giving free help
for any project (commercial or public).

regards,
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