RE: [Asterisk-Users] Toshiba Integration - MWI Light
Use a Call file to dial back to the PBX.
In voicemail.conf set the externnotify value to something like:
externnotify=/usr/local/sbin/mwi.pl
where the perl script creates the Call file. I set up a specific group and
dedicated a port to making these calls instead of chancing the glare with
the pbx. Also, my specific pbx needed some delay between dialing a feature
access key "#" and the MWI dial code itself so that is why my string is
"#www91$ext".
Here is the perl script:
#!/usr/bin/perl
my ($context,$ext,$msgs,@junk) = @ARGV;
my $tmpcallpath = "/var/tmp";
my $astpath = "/var/spool/asterisk/outgoing";
my $tmpname = "mwi-" . time();
my $tmpcallfile = "$tmpcallpath/$tmpname";
my $callfile = "$astpath/$tmpname";
$ext =~ s/[EMAIL PROTECTED]//;
if ($msgs > 0) {
$channel = "Zap/g3/#www91$ext";
} else {
$channel = "Zap/g3/#www90$ext";
}
sleep 2;
print STDERR "channel: $channel\n";
open (CALLFILE,">$tmpcallfile");
print CALLFILE qq(
Channel: $channel
MaxRetries: 0
WaitTime: 5
Context: mwi
Extension: s
Priority: 1
);
close(CALLFILE);
rename($tmpcallfile, $callfile);
Good luck!
Karl Putz
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Matthew
>Drobnak
>Sent: Wednesday, July 27, 2005 4:58 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] Toshiba Integration - MWI Light
>
>
>Hi All,
>
>On our Toshiba PBX, to light the MWI, one dials "#63__EXTENSION__" --
>how is it possible to easily trigger this after a voicemail is sent?
>
>Thanks,
>
>-Matthew Drobnak
>___
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RE: [Asterisk-Users] Put a wait in a .call file.
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Ronan
>Eckelberry
>Sent: Thursday, May 05, 2005 9:09 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Put a wait in a .call file.
>
>
>No go. Now, it picks up the phone, waits, and still only dials 6 of the
>7 #'s. It is VERY wierd
This sounds like it might be a dialplan or zapata.conf problem.
Be sure that you do not have a "stripmsd=1" in your zapata.conf (or any
value greater than 0).
In extensions.conf be sure that in the context that you are dialing from you
are not stripping digits in
the Dial command. i.e. if the Dial command passes a number to be dialed
that includes a ":" and a number
;exten => _41X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
Above, the first 2 digits, "41", will not be part of the dialstring.
Also double check the TRUNKMSD value in extensions.conf if you are using any
of the default setup
that includes this variable. Anything other than 0 will strip digits.
Good luck,
Karl Putz
>
>-Ronan
>
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RE: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Andrew >Kohlsmith >Sent: Monday, April 25, 2005 8:04 AM >Channel banks are great; the better ones (Adit600) can do far-end >disconnect >supervision and I think pretty much all of them do dynamic impedance >adjustment, meaning they're FAR less prone to echo. Just about >anyone's FXS >modules work, but be careful with FXO modules on channel banks. >Access Bank >I and IIs do *NOT* do far-end disconnect, meaning if someone on the other >side hangs up, Asterisk won't be able to tell. Andrew, What configuration do you need to do to the Adit in order to get it to recognize FXO side disconnect? I have tried a number of different settings and can never get it to pass through to *. I am configured with 2 FXS cards and 1 FXO. My FXO card is running SW ver. 1.12 and my Mainboard is at ver. 7.0.3. I am using POTS lines from SBC. My lines are getting a loop drop as I have done some testing with Voicetronix OpenSwitch cards and they do see the disconnect. Any suggestions would be appreciated. Karl Putz > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI t1
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Steve >Edwards >Sent: Tuesday, April 05, 2005 10:30 AM >To: C F; Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI >t1 > > >No. > >But before I make a fool of myself with SBC, or dive into the code, I was >wondering if anybody is using it. I.e., does it work for anybody. I am in the same boat ... I have a PRI from XO but am not seeing any II digits either. I have a call in to them to verify that they are passing the digits but have no answer yet. I'll repost as soon as I have verification either way. Karl Putz > >On Mon, 4 Apr 2005, C F wrote: > >> You sure you getting those from your telco provider? >> >> On Apr 4, 2005 12:57 PM, Steve Edwards <[EMAIL PROTECTED]> wrote: >>> I'm not seeing anything in ANI2 (aka info digits) from my newly >>> provisioned PRI T1. >>> >>> (Info digits give you a clue as to the type of phone >originating the call >>> -- cell, pay, hospital, hotel, prison, etc.) >>> >>> Is there some configuration magic needed? >>> >>> Is anybody using this feature or is it a "work in progress?" >>> >>> Thanks in advance, >>> >>> Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST >>> Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 >>> ___ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >Thanks in advance, > >Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST >Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astcc problems
There is an error in the ASTCC makefile. It places the sound files in the
wrong directory.
You can either modify the Makefile and change the line:
SOUNDSDIR=/usr/share/asterisk/sounds
to
SOUNDSDIR=/var/lib/asterisk/sounds
and then re-install. Or simply move the files from the improper directory
to the correct one.
Karl Putz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of wassim
darwish
Sent: Saturday, April 02, 2005 7:27 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] astcc problems
i have downloaded astcc and confiugured it on web but
the problem is when a call comes by the right callerid
it gives me on CLI like this:
-- Executing DeadAGI("Zap/1-1",
"astcc.agi|01475969|s") in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/astcc.agi
Detected dry run!
AGI Environment Dump:
-- accountcode =
-- callerid = 01475969
-- calleridname = unknown
-- channel = Zap/1-1
-- context = incoming
-- dnid = unknown
-- enhanced = 0.0
-- extension = s
-- language = en
-- priority = 3
-- rdnis = unknown
-- request = astcc.agi
-- type = Zap
-- uniqueid = 1112430048.4
Apr 2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-tone does not exist in
any format
Res is
Silent Level is
Card no is 12345
Card has face value 3 and has used 0
3 dollars and 0 cents remain
Apr 2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-youhave does not exist
in any format
-- Playing 'digits/3' (language 'en')
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-dollars does not exist
in any format
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-remaining does not
exist in any format
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-badphone does not
exist in any format
-- AGI Script astcc.agi completed, returning 0
I dont know what the problem and what this warnings
mean and how can i fix them please help.
and thanks
__
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Make Yahoo! your home page
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RE: [Asterisk-Users] ASTCC - Regex: How to "Country" but "special City" different?
>my $sth = $dbh->prepare("SELECT * FROM routes WHERE " . $dbh->quote($num
>ber) . " RLIKE pattern ORDER BY LENGTH(pattern) DESC");
>
>Does it mean I just need to use:
>^61.* 100
>^6178.* 150
>^615.* 130
>^61342.* 180
Ronald,
The ASTCC sql SELECT used will return the routes entry that matches the
longest route pattern that you have entered into the table. I.e it will
return the most specific match possible.
I believe you can drop the "." and the "*" from the end of the routes
entries as these are redundant and add length to a Regex that is simply
looking to match the first N characters of the dialed number.
Your assumption is correct in your example that you can create separate
entries for specific city codes and also have a default "else" route that
rates the remainder of the country.
>
>would already work? A number of ^6154333231 would be considered as the
>first case and cost 100 pennies ???
No, the 615 would be recognized as your third entry and billed at 130.
Karl Putz
>
>
>bye
>
>Ronald
>
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RE: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Steven >Critchfield >Sent: Monday, March 07, 2005 6:08 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Question about AGI vs. FastAGI vs. >straight C/DB development > > > >You said a magic word there, customers. You need to become versed in the >ins and outs of licensing. If you write it in C and distribute it, you >must also offer to those you distribute asterisk your modifications and >linked in code. > >For AGI and FastAGI, it isn't linked to GPL code in asterisk and >therefore is sheltered from the GPL license of asterisk. But then you >need to verify you aren't using any libraries that are distributed under >a GPL license or you would also have to open source your code upon >distribution. Please correct me if I'm wrong here but, the script/process called by the AGI could be any executable adhering to the AGI standard. This could be a compiled C or any language executable that simply parses the startup info provided by the AGI and communicates with * via stdin and stdout. This means that the executable would be standalone and not need to be linked to the GPL code. I think this means that the source of any AGI application would not be subject to the GPL licensing if the author would so choose. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stutter Tone
-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Steven >Critchfield >Sent: Friday, March 04, 2005 10:26 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Stutter Tone > > >On Fri, 2005-03-04 at 21:10 -0600, Anton Krall wrote: >> I think I have something misconfigured regarding voicemails. They work >> great, I have this setup: >> >> Sip.conf >> >> [ext1] >> Context=phones >> Mailbox=201 >> >> Voicemail.conf >> >> [home] >> >> 201,password,name,[EMAIL PROTECTED] >> >> Voicemail delivery and all works great but when I check sip >extension ext1 >> (analog phone using a Granstream ATA 286), the stutter tone signaling >> message waiting does not work. > >SIP dialtones come from the SIP device. Look up the config on your SIP Steven, I think he is referring to MWI notification to the ATA. Anton - I believe you need to identify the voicemail context in the sip.conf if you are using something other than [default] like: [EMAIL PROTECTED] Karl Putz >device. >-- >Steven Critchfield <[EMAIL PROTECTED]> > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC questions
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Ronald
>Wiplinger
>Sent: Thursday, March 03, 2005 2:47 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] ASTCC questions
>
>
>Ronald Wiplinger wrote:
>
>(Correcting my own message)
>
>> I have setup ASTCC as:
>>
>> trunk:
>>
>> NuFone IAX2 NuFone
>
>should be:
>NuFone IAX2 [EMAIL PROTECTED] !!!
>
>
>1. So far I can call out, but I cannot call in. - Any hints?
>2. ASTCC shows me for my test calls only:
>In Cards that I used from 1 60 pennies
>If I try to get detail info from the card, than I get:
>
>/Card /*886228803959 */ has used /*60* of *10* units
>
>Caller*ID Called Number Trunk Disposition Billable Seconds
>Billed Cost
>
>
>but no detail data!!!Any hints???
Ronald,
The CVS ASTCC has an error in the database table structure for the call
records.
See http://bugs.digium.com/bug_view_page.php?bug_id=0002796
for a patch to the cgi scripts that create the table. Basically, the
"callstart" field is missing in the
CREATE table cdrs statement.
The above link also has a few additions to ASTCC that may be interesting to
you. Specifically,
there is an extension that allows you to use the caller id as the account
number but also require a
PIN to complete the call.
Karl Putz
>
>
>
>bye
>
>Ronald
>
>>
>> routes:
>>
>> ^1415.* California NuFone 0 0 200
>>
>> iax.conf
>> =
>> register => User:[EMAIL PROTECTED]
>>
>> [NuFone]
>> type=peer
>> host=switch-1.nufone.net
>> secret=my_secret
>>
>> [NuFone]
>> type=user
>> secret=my_secret
>> context=fromNuFone
>>
>>
>> extensions.conf
>> ==
>> [NuFone]
>> exten =>
>> _91NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
>> exten => _9011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
>>
>>
>>
>>
>> With above settings I see in CLI> when I am dialing:
>>-- Executing NoOp("SIP/886228803959-1e6d", "SetCallerID()") in new
>> stack
>>-- Executing Dial("SIP/886228803959-1e6d",
>> "IAX2/[EMAIL PROTECTED]/14159625000") in new stack
>>-- Called [EMAIL PROTECTED]/14159625000
>>-- Call accepted by 66.225.202.72 (format ulaw)
>>-- Format for call is ulaw
>>-- IAX2/NuFone-11 answered SIP/886228803959-1e6d
>>-- Hungup 'IAX2/NuFone-11'
>> == Spawn extension (VoIP_customer_Phone, 914159625000, 2) exited
>> non-zero on 'SIP/886228803959-1e6d'
>>
>> It works !!!
>>
>>
>> Changing the settings in extensions.conf to:
>>
>> [NuFone]
>> ;exten =>
>> _91NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
>> ;exten => _9011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
>> ;
>> exten => _91NXXNXX,1,NoOp(SetCallerID(${username}))
>> exten =>
>> _91NXXNXX,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}})
>> exten => _91NXXNXX,3,hangup
>> ;
>> exten => _9011N.,1,NoOp(SetCallerID(${username}))
>> exten => _9011N.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}})
>> exten => _9011N.,3,hangup
>>
>>
>>
>>
>> gives me in CLI> by redialing the same number:
>>
>>
>>
>>-- Executing NoOp("SIP/886228803959-e043", "SetCallerID()") in new
>> stack
>>-- Executing DeadAGI("SIP/886228803959-e043",
>> "astcc.agi|886228803959|14159625000") in new stack
>>-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
>>-- Playing 'digits/10' (language 'en')
>>-- Registered IAX2 to '69.73.19.178', who sees us as
>> 61.220.121.20:4569
>>-- Playing 'digits/2' (language 'en')
>>-- AGI Script Executing Application: (DIAL) Options:
>> (IAX2/NuFone/14159625000|30|HL(1998:6:3))
>>-- Limit Data:
>>-- timelimit=1998
>>-- play_warning=6
>>-- play_to_caller=yes
>>-- play_to_callee=no
>>-- warning_freq=3
>>-- start_sound=UNDEF
>>-- warning_sound=timeleft
>>-- end_sound=UNDEF
>>-- Called NuFone/14159625000
>> Mar 3 14:00:31 WARNING[8102]: chan_iax2.c:6280 socket_read: Call
>> rejected by 66.225.202.72: No such context/extension
>>-- Hungup 'IAX2/NuFone-3'
>> == No one is available to answer at this time (1:0/0/0)
>>-- AGI Script astcc.agi completed, returning 0
>>-- Executing Hangup("SIP/886228803959-e043", "") in new stack
>> == Spawn extension (VoIP_customer_Phone, 914159625000, 3) exited
>> non-zero on 'SIP/886228803959-e043'
>>
>>
>>
>> Why it tells me: No such context/extension ???
>>
>> What do I need to change?
>>
>> Thanks for your help in advance!
>>
>>
>> bye
>>
>> Ronald
>>
>>
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>>
>>
>
>
>--
>Ronald Wiplinger (CEO of ELMIT)
>http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208
>- I'm a SpamCon Foundation Member, #694, Verify it at
>
RE: [Asterisk-Users] /dev/zap not created
>I'm not 100% sure, but I think Fedora Core 2 uses UDEV. Look through >the output of ps -A and see if there is a udevd running. If there is >you're running udev and need to read README.udev which is in the zaptel >source directory. > > I'm running FC2 V2.6.10-1.14 and it is not using udev. When you build zaptel, follow the make install with make devicesthis will create the /dev/zap devices and possibly make config this will place the startup scripts for the zaptel service into the rc.d area Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home scary log
Please all keep in mind that there are plenty of additional configs possible to Iptables. I should have restricted the originating IP address for TCP port 22 to come from at least my dhcp served address range. That would have blocked all hackers except those originating from within my specific ISP's dhcp served range. Not perfect but a good sight better that wide open! Karl Putz >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Christian >Moller >Sent: Thursday, February 10, 2005 11:03 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >Hi, >OK, well, I've disabled SSH/HTTP already so lets hope I will have >my system >working! >Best and thanks, >Christian > > >- Original Message - >From: "Karl H. Putz" <[EMAIL PROTECTED]> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >Sent: Thursday, February 10, 2005 4:56 PM >Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >>I had the system setup to allow http and ssh. >> >> The hack came in through ssh. >> >>>-Original Message- >>>From: [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] Behalf Of Christian >>>Moller >>>Sent: Thursday, February 10, 2005 10:39 AM >>>To: Asterisk Users Mailing List - Non-Commercial Discussion >>>Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log >>> >>> >>>Hi, >>>I've also been a little worried about the security. How did they >>>connect to >>>your system? Through telnet or what? >>>Since I've disabled all such services. >>>Best, >>>Christian >>> >>> >>>- Original Message - >>>From: "Karl H. Putz" <[EMAIL PROTECTED]> >>>To: "Jean-Louis curty" <[EMAIL PROTECTED]>; "Asterisk Users >Mailing List - >>>Non-Commercial Discussion" >>>Sent: Thursday, February 10, 2005 4:18 PM >>>Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log >>> >>> >>>> You've likely been hacked. >>>> >>>> I have recently had a similar incident where a hacker guessed my root >>>> password (MY BAD) and set up an ebay password skimming site. >>>> >>>> I noticed it when I got similar non-deliverable email messages. >>>> >>>> Obviously, first change your password and then look at the >/var/www/html >>>> directory and see if there are unwelcome pages there. Also be sure to >>>> check >>>> who is logged in currently. I caught the (*%#@ SOB logged in and >>>> bounced >>>> the bastard. >>>> >>>> For what it's worth, the hacker's IP address was: 81.12.141.150. >>>> >>>> >>>> Karl Putz >>>> >>>>>-Original Message- >>>>>From: [EMAIL PROTECTED] >>>>>[mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis >>>>>curty >>>>>Sent: Thursday, February 10, 2005 9:10 AM >>>>>To: Asterisk Users Mailing List - Non-Commercial Discussion >>>>>Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log >>>>> >>>>> >>>>>Hi everybody, >>>>> >>>>>I'm testing [EMAIL PROTECTED] 0.4, >>>>>looks great so far >>>>> >>>>>I was working when I have been alerted by a bip comming from >the * pc... >>>>> >>>>>I connected a screen to it and saw that there was a message which >>>>>looked like : >>>>> >>>>> >>>>>Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... >>>>>asterisk1 >>>>> >>>>> >>>>> >>>>>so I stopped asterisk, type mail and got a strange mail saying that >>>>>user [EMAIL PROTECTED] could not be reached and body was like if it was >>>>>the result of commands ifconfig etc >>>>> >>>>>unfortunally I don't have the message anymore but I went to the log >>>>> >>>>>and saw this >>>>>Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: >>>>>from=<[EMAIL PROTECTED]>, size=329, class=0, nrcpts=1, >>>>>msgid=<[EMAIL PROTECTED]>, proto=ESMTP, >>>>>daemon=MTA, relay=asterisk1.local [127.0.0
RE: [Asterisk-Users] asterisk@home scary log
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Steven >Critchfield >Sent: Thursday, February 10, 2005 11:47 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >On Thu, 2005-02-10 at 10:56 -0500, Karl H. Putz wrote: >> I had the system setup to allow http and ssh. >> >> The hack came in through ssh. > >I doubt you where hacked via ssh. Most likely you had your password >brute force cracked. That is what I meant to report to the list. SSH was simply the transport mechanism. Karl > >> >-Original Message- >> >[mailto:[EMAIL PROTECTED] Behalf Of Christian >> >Moller >> >Sent: Thursday, February 10, 2005 10:39 AM >> >Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log >> >your system? Through telnet or what? > >What moron still uses telnet these days? > >> >- Original Message - >> >From: "Karl H. Putz" <[EMAIL PROTECTED]> >> >Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log >> > >> > >> >> You've likely been hacked. >> >> >> >> I have recently had a similar incident where a hacker guessed my root >> >> password (MY BAD) and set up an ebay password skimming site. > >This is a good example of why ease of use is not always a good thing. >Had you actually had to learn more before you had an install, you would >have been through a text or two that mention password strengths. > >And not to disparage the creator/maintainer of [EMAIL PROTECTED], but you >really need to trust that your install was a little hardened before >placing it on the network. >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home scary log
I had the system setup to allow http and ssh. The hack came in through ssh. >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Christian >Moller >Sent: Thursday, February 10, 2005 10:39 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >Hi, >I've also been a little worried about the security. How did they >connect to >your system? Through telnet or what? >Since I've disabled all such services. >Best, >Christian > > >- Original Message - >From: "Karl H. Putz" <[EMAIL PROTECTED]> >To: "Jean-Louis curty" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - >Non-Commercial Discussion" >Sent: Thursday, February 10, 2005 4:18 PM >Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >> You've likely been hacked. >> >> I have recently had a similar incident where a hacker guessed my root >> password (MY BAD) and set up an ebay password skimming site. >> >> I noticed it when I got similar non-deliverable email messages. >> >> Obviously, first change your password and then look at the /var/www/html >> directory and see if there are unwelcome pages there. Also be sure to >> check >> who is logged in currently. I caught the (*%#@ SOB logged in and bounced >> the bastard. >> >> For what it's worth, the hacker's IP address was: 81.12.141.150. >> >> >> Karl Putz >> >>>-Original Message- >>>From: [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis >>>curty >>>Sent: Thursday, February 10, 2005 9:10 AM >>>To: Asterisk Users Mailing List - Non-Commercial Discussion >>>Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log >>> >>> >>>Hi everybody, >>> >>>I'm testing [EMAIL PROTECTED] 0.4, >>>looks great so far >>> >>>I was working when I have been alerted by a bip comming from the * pc... >>> >>>I connected a screen to it and saw that there was a message which >>>looked like : >>> >>> >>>Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... >>>asterisk1 >>> >>> >>> >>>so I stopped asterisk, type mail and got a strange mail saying that >>>user [EMAIL PROTECTED] could not be reached and body was like if it was >>>the result of commands ifconfig etc >>> >>>unfortunally I don't have the message anymore but I went to the log >>> >>>and saw this >>>Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: >>>from=<[EMAIL PROTECTED]>, size=329, class=0, nrcpts=1, >>>msgid=<[EMAIL PROTECTED]>, proto=ESMTP, >>>daemon=MTA, relay=asterisk1.local [127.0.0.1] >>>Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: >>>[EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, >>>xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] >>>[127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for >>>delivery) >>>Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: >>>[EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, >>>xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] >>>[127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for >>>delivery) >>>Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: >>>to=<[EMAIL PROTECTED]>, ctladdr=<[EMAIL PROTECTED]> (0/0), >>>delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, >>>relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK >>>1107998984) >>>Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: >>>to=<[EMAIL PROTECTED]>, ctladdr=<[EMAIL PROTECTED]> (0/0), >>>delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, >>>relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK >>>1107998984) >>> >>> >>>the thing is i did not send any message to [EMAIL PROTECTED] nor to >>>somebody at yahoo, >>> >>> >>>anybody got the same ? what can I do ?? >>> >>>thanks >>>jl >>>___ >>>Asterisk-Users mailing list >>>Asterisk-Users@lists.digium.com >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home scary log
You've likely been hacked. I have recently had a similar incident where a hacker guessed my root password (MY BAD) and set up an ebay password skimming site. I noticed it when I got similar non-deliverable email messages. Obviously, first change your password and then look at the /var/www/html directory and see if there are unwelcome pages there. Also be sure to check who is logged in currently. I caught the (*%#@ SOB logged in and bounced the bastard. For what it's worth, the hacker's IP address was: 81.12.141.150. Karl Putz >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Jean-Louis >curty >Sent: Thursday, February 10, 2005 9:10 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] [EMAIL PROTECTED] scary log > > >Hi everybody, > >I'm testing [EMAIL PROTECTED] 0.4, >looks great so far > >I was working when I have been alerted by a bip comming from the * pc... > >I connected a screen to it and saw that there was a message which >looked like : > > >Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... >asterisk1 > > > >so I stopped asterisk, type mail and got a strange mail saying that >user [EMAIL PROTECTED] could not be reached and body was like if it was >the result of commands ifconfig etc > >unfortunally I don't have the message anymore but I went to the log > >and saw this >Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: >from=<[EMAIL PROTECTED]>, size=329, class=0, nrcpts=1, >msgid=<[EMAIL PROTECTED]>, proto=ESMTP, >daemon=MTA, relay=asterisk1.local [127.0.0.1] >Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: >[EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, >xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] >[127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for >delivery) >Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: >[EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, >xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] >[127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for >delivery) >Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: >to=<[EMAIL PROTECTED]>, ctladdr=<[EMAIL PROTECTED]> (0/0), >delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, >relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK >1107998984) >Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: >to=<[EMAIL PROTECTED]>, ctladdr=<[EMAIL PROTECTED]> (0/0), >delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, >relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK >1107998984) > > >the thing is i did not send any message to [EMAIL PROTECTED] nor to >somebody at yahoo, > > >anybody got the same ? what can I do ?? > >thanks >jl >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sample REGEX's for astcc
remember to use the ^ to indicate matching at the beginning of the number. i.e ^01144 should be all you need to match any international call going to country code 44. Karl Putz >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Jason >Kawakami >Sent: Wednesday, February 09, 2005 7:05 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] sample REGEX's for astcc > > >So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that >should match NXXNX. Right? > >I built another route 01144[0-9]* that I thought would match 01144X. and >send the call to the UK but the script is matching 01144207108 With the >first route. > >Can someone smarter than me help with some samples? Please? If I can get >one for 1NXXN. and 01144. I should be able to figure the rest out. > >Jason Kawakami >www.optellabs.com >Salt Lake City, UT > > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astcc with multiple access
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Mike Nugent >Sent: Tuesday, February 08, 2005 5:31 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] astcc with multiple access > > > > >I'm looking at astcc and it seems that setting up a script that will >allow multiple people to access a calling card simultaneously would be >fairly difficult. Before I endevour to develop this, has anyone already >done it/looked at this/can point me in the right direction? > >Thanks. > >-- >Mike Nugent <[EMAIL PROTECTED]> > > Mike, It would not really be very difficult to remove all references to the 'inuse' field. But, you may want to consider possibly using that field as a counter rather than a boolean flag. That way, you could impose a maximum number of simultaneous users of an account. Basically rather than setting 0 or 1, you could increment the field by 1 when a new user authenticates and decrement when that caller drops. Check the value against your desired maximum instead of a simple boolean 1 to block additional users of that account. Be aware that if you allow this type of use, your account balance could very easily go negative. Good luck! Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC simultenous calls per card
>-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of thieumS >Sent: Tuesday, February 08, 2005 10:09 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] ASTCC simultenous calls per card > > >Hi guys, >do you know if it's possible to handle more than 1 call per card >with astcc ? Astcc allows only a single simultaneous call per account number. Any pre-paid application will have some inherent problems in allowing multiple users/callers accessing the same pool of funds. Especially if the amount remaining in the account could be entirely used up by a single caller. Karl Putz > >Thank you. > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC error on free calls
Trevor,
The problem is a "division by zero" issue in the astcc.agi @ line 538 (I
have made a few mods so my line #'s may not be exactly the same). The line
reads
:
$maxmins = int(($credit - $adjconn) / $adjcost);
you may want to change the script to something like:
if ($adjcost < 1) {
$maxmins = ;
} else {
$maxmins = int(($credit - $adjconn) / $adjcost);
}
I set $maxmins = $credit on my system and that will give 1 minute per penny
of account balance as the max duration of the call but will not charge any
duration related fees against the account. Any connection fees would still
be assessed at call termination, though.
Karl Putz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Trevor G.
Hammonds
Sent: Saturday, February 05, 2005 12:03 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] ASTCC error on free calls
I set up certain routes in my ASTCC application to be free of charge. When
a user attempts to dial one of these numbers, the announcement plays the
prompts "This call will cost", "nothing", and then terminates the script,
dropping the call, leaving the card locked in the database as being in use.
Any ideas?
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] ASTCC Apllication
>The SQL statement in astcc returns all the matched patterns with the >longest, most specific match first and uses only that first match in its >processing. So you could also use the pattern: '.' to match any dialed >number not already matched as a default BUT BE SURE to set that cost high >enough to cover yourself. > > Sorry all for replying to my own posting but I realized that my advise on the above "default" pattern was a BAD idea. The main issue is that it does not allow for blocking any routes especially to 1-900 or other pay exchanges. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC Apllication
-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa >Sent: Friday, February 04, 2005 4:50 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] ASTCC Apllication > > >Hello, >I have some problem using ASTCC application. I've installed the application and everything works >well. I've created card numbers, routes trunk and others. When I dial the desired number (77) in >my case, I'm prompted to enter my card number. All goes well till I'm prompted to enter the >destination number. When I enter a destination number, the system says it's not a recognized >number and the call doesn't go through. Can any one help me out with this issue? Is there a file >where I can define extensions like in extensions.conf? Daniel, It sounds like the problem is the pattern you are trying to use in the "routes" table. The pattern should be a REGEX for matching the dialed number to the appropriate cost for that call. Take a look at http://dev.mysql.com/doc/mysql/en/pattern-matching.html for more specifics on MySQL REGEX matching. In the US, for example, I would use the pattern: '^1312' to match for calls to Chicago or '^01149' for calls to Germany. You can also match for city codes or especially Cellular "exchanges" in specific countries where the termination costs are much higher than land-line termination. The SQL statement in astcc returns all the matched patterns with the longest, most specific match first and uses only that first match in its processing. So you could also use the pattern: '.' to match any dialed number not already matched as a default BUT BE SURE to set that cost high enough to cover yourself. Good luck! Karl Putz > > >Thanks. > >Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astcc digit timeout
> El Jue 03 Feb 2005 14:02, Steve Totaro escribió:
> > does anyone know how to change the timeout on digit entry in
> astcc. if you
> > call the app and start entering a pin, you have about 2 seconds to enter
> > the next number or you get timed out. i cannot find any info
> on this from
> > the lists or google.
Steve,
modify the agi to add the specific inter-digit timeout in milliseconds you
would like after the prompt filename in the get_data calls.
i.e. use: $cardno = $AGI->get_data("astcc-accountnum",5000);
if you want a 5 second allowable delay between digits.
Karl Putz
> >
> > Thanks,
> > Steve
>
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RE: [Asterisk-Users] How to use ASTCC with SIP ??
The current astcc Makefile puts the sound files into the wrong directory.
It uses /usr/share/asterisk/sounds but it should be
/var/lib/asterisk/sounds.
Karl Putz
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
> Sent: Saturday, January 29, 2005 12:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] How to use ASTCC with SIP ??
>
>
>
> I got this error when i try to dial:
>
> -- Executing Answer("SIP/8000104-71a3", "") in new stack
> -- Executing DeadAGI("SIP/8000104-71a3", "astcc.agi") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
> astcc-tone does not exist in any format
> Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
> astcc-accountnum does not exist in any format
> Jan 29 18:11:37 WARNING[3412]: file.c:779 ast_streamfile: Unable to open
> astcc-accountnum (format alaw): No such file or directory
> == Spawn extension (prepaid, 77, 2) exited non-zero on
> 'SIP/8000104-71a3'
>
> Can somebody tell me why and how to solve it ??
>
> Regards.
> Daniel.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Darren
> Wiebe
> Sent: samedi 29 janvier 2005 18:12
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How to use ASTCC with SIP ??
>
> I would recommend using the local trunk and then you just need a context
>
> that will dial out in your extensions.conf. Just put the context name
> into the "Peer/Trunk" field on the trunks page. Currently there is not
> support in astcc for oh-323. It would be trivial to add but
>
> Darren Wiebe
> [EMAIL PROTECTED]
>
> Daniel Eboa wrote:
>
> > Hello List,
> >
> > I've set up asterisk and install astcc application, everything was
> > well installed, but i'm having problem using astcc with SIP. I don't
> > have any Trunk card or any other analogic VoIP card connected to my
> > asterisk box. I'm using SIP and asterisk-oh323 to connect to my VoIP
> > provider. Does anyone knows how I can use astcc to work with my config
> ?
> >
> > Thanks.
> >
> >---
> -
> >___
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RE: [Asterisk-Users] VoIP-to-TDM processing on-card?
Sangoma is planning to release a multi-port (T1/E1) card later this year with DSP resources available. Karl Putz > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Michael > Baird > Sent: Thursday, January 20, 2005 2:42 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] VoIP-to-TDM processing on-card? > > > On Thu, 2005-01-20 at 14:05 -0500, Olson, Dana wrote: > > I did look there. If you read my follow up, I screwed up the > original question. What I want is a card with multiple T1 ports > that do the processing on the card, and not on the system CPU. > > > > I'm not aware of any cards with DSP's on board for Asterisk (nice > thought), the Digium cards I have rely on the PC's CPU to handle the > calls. > > > Is there a mailing list for Asterisk where people treat each > other in a civil manner? > > __ > > Dana Olson > > It's only one guy who seems to attack each poster for not posting in a > manner of which he approves (there is one/two of these fellows on every > mailing list), don't let him ruin your day, this list is quite helpful > and many guys will give you a good answer without the extra attitude. > > Regards > Michael Baird > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI info digits question
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Michael K. > Rodriguez User > Sent: Thursday, January 20, 2005 2:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] PRI info digits question > > > Does anyone know how does asterisk handles INFO digit from a PRI line? > Can info digit be used in extensions.conf to signal a call from a public > phone? I believe that the II digits are only available when using the CVS Head branch of Asterisk. There is a variable CALLINGANI2 that gets populated with what should be the info digits. I have been working on getting a PRI up and running but have not yet had a chance to verify this yet. Karl Putz > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tony > Mountifield > Sent: Friday, January 14, 2005 12:56 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 > server > > > In article <[EMAIL PROTECTED]>, > Matt Riddell <[EMAIL PROTECTED]> wrote: > > Steve Hanselman wrote: > > > I'm assuming that other non Digium cards work in it, but yes, > you're right. > > > > > > Has anybody run any other PCI cards in those slots under > Linux and seen > > > interrupts from those cards? I am not using the HP-Compaq server listed but am having the Exact problems mentioned on a Dual EMT64 server from Tatung -All outward appearance is that the TE410 is working (module loads, ztcfg runs with no error, etc) except no interrupts. I successfully ran a T100P card in the same slot using the wct1xxp module compiled from the same CVS Head source as the non-working wct4xxp module. Karl Putz > > > > You'd be hard pressed to find a standard card requiring accurate > > interrupts 1000 times per second... > > That's not the issue - the issue is: can another type of card in the > same slot raise an interrupt when it needs to? > > I had the same problem of no interrupts from a TE405P in a 1U chassis > with a backplane. When we demonstrated that a standard LAN card was > also unable to generate an interrupt, the vendor accepted that it could > be a problem with their backplane, and came up with a hardware fix. > > Cheers > Tony > -- > Tony Mountifield > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC configuration problem
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steve > Totaro > Sent: Wednesday, January 12, 2005 12:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] ASTCC configuration problem > > > > > > > > If nothing else, my efforts are documented for anyone else in the > > > > same boat. > > > > It seems that you can debug agi by typing agi debug at the * command > > line. > > > > Amazing! Here is the output. I am assuming that since astcc-tone > didnt > > > > play, the problem lies there. Thoughts? > > > > > > Steve, > > > > > > I just remembered... > > > > > > The Makefile puts all its sounds into /usr/share/asterisk/sounds but * > > needs > > > the sounds in /var/lib/asterisk/sounds. > > > > > > > > > Karl > > > > Karl, > > > > Thanks alot! Everything is working now but I dont get any > audio in either > > direction. If I dial the same external number from the same extension, > > everything is fine but going through the AGI breaks the audio. > > To clarify, I get audio just fine while entering the pin and phone number > but when the call connects over my IAX provider there is no audio > in either > direction. Steve, I sounds like the problem may be in your "trunks" table definition of the IAX2 connection. Unfortunately, I am only using TDM interfaces so I do not know what the path field should contain for IAX. For Zap interfaces, I simply put the group I want to use. I.e. "g2". Karl > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC configuration problem
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> Totaro
> Sent: Wednesday, January 12, 2005 10:30 AM
>
> If nothing else, my efforts are documented for anyone else in the
> same boat.
> It seems that you can debug agi by typing agi debug at the * command line.
> Amazing! Here is the output. I am assuming that since astcc-tone didnt
> play, the problem lies there. Thoughts?
Steve,
I just remembered...
The Makefile puts all its sounds into /usr/share/asterisk/sounds but * needs
the sounds in /var/lib/asterisk/sounds.
Karl
>
> AGI Debugging Enabled
> -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024,
> actual format = 2
> -- Executing Answer("IAX2/[EMAIL PROTECTED]/5", "") in new stack
> -- Executing DeadAGI("IAX2/[EMAIL PROTECTED]/5", "astcc.agi") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> AGI Tx >> agi_request: astcc.agi
> AGI Tx >> agi_channel: IAX2/[EMAIL PROTECTED]/5
> AGI Tx >> agi_language: en
> AGI Tx >> agi_type: IAX2
> AGI Tx >> agi_uniqueid: 1105546846.5
> AGI Tx >> agi_callerid: "(Softphone) Steve\'s Dell" <200>
> AGI Tx >> agi_dnid: 77
> AGI Tx >> agi_rdnis: unknown
> AGI Tx >> agi_context: from-internal
> AGI Tx >> agi_extension: 77
> AGI Tx >> agi_priority: 2
> AGI Tx >> agi_enhanced: 0.0
> AGI Tx >> agi_accountcode:
> AGI Tx >>
> AGI Rx << ANSWER
> AGI Tx >> 200 result=0
> AGI Rx << STREAM FILE astcc-tone 0123456789
> AGI Tx >> 200 result=0 endpos=0
>
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RE: [Asterisk-Users] ASTCC configuration problem
Steve, What version of MySQL are you running? I upgraded to 4.1.8 and ran into the problem below. I initially tested with the user root and the default blank password and was OK. But when I changed over to a new user with a password, I noticed an error message in the httpd logs: Client does not support authentication protocol requested by server; consider upgrading MySQL client Once I converted to OLD_PASSWORD for the user, I was OK. Remember to set the password for the [EMAIL PROTECTED] matching the host name you have used in the config file. http://dev.mysql.com/doc/mysql/en/Old_client.html Karl Putz > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steve > Totaro > Sent: Tuesday, January 11, 2005 6:39 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] ASTCC configuration problem > > > Same problem. I dropped all the tables and ran the above. I also checked > the sockets and they are correct. Any other ideas? > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC configuration problem
> CREATE TABLE `cdrs` ( > > `cardnum` char(40) default NULL, > > `callerid` char(80) default NULL, > > `callednum` char(80) default NULL, > > `trunk` char(40) default NULL, > > `disposition` char(20) default NULL, > > `billseconds` int(11) default NULL, > > `billcost` int(11) default NULL > ) TYPE=MyISAM; The cdrs table creation script is missing the callstart field. I use the sql below: CREATE TABLE `cdrs` ( `cardnum` varchar(40) default NULL, `callerid` varchar(80) default NULL, `callednum` varchar(80) default NULL, `trunk` varchar(40) default NULL, `disposition` varchar(20) default NULL, `billseconds` int(11) default NULL, `billcost` int(11) default NULL, `callstart` varchar(24) ) TYPE=MyISAM; Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC - error on call end
There is a field missing in the admin.cgi CREATE for cdrs. add: callstart CHAR(24) to the cdrs table There is a patch to fix the cgi at http://bugs.digium.com/bug_view_page.php?bug_id=0002796 It just hasn't made it through to CVS yet. Karl Putz > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: Tuesday, January 11, 2005 9:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] ASTCC - error on call end > > > Hi > > I get an error popping up when the call ends as follows: > > > DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at > /var/lib/asterisk/agi-bin/astcc.agi line 90, line 32. > > Does anyone else get this same thing? > Looks as if my database table is wrong, or something else is > up...not sure > > Thanks > Clive > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
I have been having this exact problem with a Tatung dual EMT-64 server as well. I have been trying to get a TE410P running and all looks great, driver loads, runs ztcfg OK, etc. but no interrupts are ever processed. One additional piece of info that I have not seen in this thread is that I am able to successfully start and run a T100P card in this system. In the same PCI slot, wct1xxp driver built from the same CVS HEAD version as the wct4xxp. Just hoping this might shed some light on the problem for any Digium folks monitoring the forum. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
> > I really am at my wits end about this one. Some people report this > > card and server working fine while others (like myself) can't get it > > going no matter what. I have been told by the Digium distributor in > > our country that this card simply "not compatible with some > > motherboards". Sounds very weak for such an expemsive card > > I have a 410 working in a DL360 fine (FC2) - had to get the right zaptel > config or the system would crash, but once that was sorted all went > fine. > > > Steve > I have been trying to get a span up from XO on a DL360 running Whitebox using the 410 card. The system had been acting normally but I was constantly in a Red alarm state. I had been using the Stable V1.0 branch of all sources having done a CVS update as late as 01/03/05. I went ahead and removed all libraries, executables, and modules and downloaded the current development branch from CVS last evening and my link is now established without alarm. The updates to libpri and zaptel in the CVS Head may well address other difficulties with newer systems/motherboards. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: Monday, January 03, 2005 1:35 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server > > > Hi, > > Has anyone had success using a TE410P card in an HP-Compaq DL380 G4 > server? > > For me the card is detected fine, but the system just never sees an > interrupt from the card. I've tried everything I can think of. The card > definitely works. > > Its Fedora Core, but we also tried a stock 2.6.10 kernel. We tried with > and without Hyperthreading, with "noapic", we disabled all the > hardware we > could, etc etc... > > So - has anyone used this combination successfully? > > This customer has quite a thing for HP - so if the DL380 is going to be a > problem, is there another rack-mount HP server that someone can vouch for > with the TE410P? I am using DL360 G3s with dual Xeon processors (Hyperthreading turned on) with no problems. I am running a Whitebox 2.4 kernel. Karl Putz > > Thanks, > Steve > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC gsm files
The AGI looks in /var/lib/asterisk/sounds for its sound files with names not fully qualified. This looks like a bug in the ASTCC makefile or the astcc.agi should refer to the soundfiles in the /usr/share/asterisk/sounds directory explicitly. Karl Putz -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of mohammadSent: Saturday, January 01, 2005 9:45 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] ASTCC gsm files Hi ALL; I installed ASTCC, but the sound files(*.gsm) cannot be opened.All of them are located in : /usr/share/asterisk/sound. Asterisk says: Jan 2 06:01:54 WARNING[9759]: file.c:475 ast_openstream: File astcc-tone does not exist in any formatJan 2 06:01:54 WARNING[9759]: res_agi.c:436 handle_streamfile: Unable to open astcc-tone Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC in production
-Original Message- >From: Darren Wiebe [mailto:[EMAIL PROTECTED] >Sent: Friday, December 17, 2004 6:55 PM >To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] ASTCC in production > > >No there is not. That would probably be easy enough to code in but I'm >not aware of it being done yet. > I was planning on adding the info digit parsing to the AGI but I am more interested in a good base Asterisk starting point. I had been focusing on the most current Stable release but since the updates for II are not in that branch, I need to move to CVS Head. I was hoping for a suggestion of a specific date tag to retrieve from CVS that has been tested enough to use in a production system. Preferably a version that 1 or more list users are actually using in a production/business environment > >Darren Wiebe >[EMAIL PROTECTED] >>Karl H. Putz wrote: >> I am looking for the most stable version of Asterisk to use with ASTCC >> for a production environment. >> >> It does not appear that any of the Stable versions will be suitable >> since they do not support US PRI ANI Info digit collection and hence >> could not apply surcharges for payphone use, etc. >> >> Is there a specific CVS Head version date that includes the II updates >> and has proven to be stable enough to run in production? >> >> Thanks for the help. >> >> >> Karl Putz >> >> >> >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users >> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC in production
I am looking for the most stable version of Asterisk to use with ASTCC for a production environment. It does not appear that any of the Stable versions will be suitable since they do not support US PRI ANI Info digit collection and hence could not apply surcharges for payphone use, etc. Is there a specific CVS Head version date that includes the II updates and has proven to be stable enough to run in production? Thanks for the help. Karl Putz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and ast_data
Gunnar, Add the voicemail options you want to use for that mailbox to the "users" table as text with multiple option=value pairs separated by a '|' (pipe) character just as it was in the voicemail.conf file. I believe the envelope=yes option is what you want to set. Karl Putz Forte Communications -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gunnar Schaller Sent: Thursday, October 21, 2004 10:41 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Voicemail and ast_data Hi, I have a problem with Voicemail. Asterisk 1.0.1 patched with ast_data connected to a mysql-server, mailboxes are in a mysql database. When I call to VoicemailMain to hear my messages it don't tell me the time the message was left. Only "Message 1" and then the message. For testing I defined a mailbox in voicemail.conf. Hearing messages from this account in VoicemailMain tells me the time the message was left. For example "Message one, received thuesday ." and then the message. There are no special options I set. Can anybody give me a hint? Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Realtime config and MWI
Currently setting mailbox= in sip.conf along with appropriate additional info is required to set and clear MWI for sip clients. MySQL peer table does not include the mailbox variable and, while ast_data does include the mailbox variable, the polling architecture of chan_sip does not currently work with it to send the Notify message for MWI. I was wondering if the new RealTime config functions would resolve this issue. Karl -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Boehm Sent: Monday, October 18, 2004 11:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New Realtime config and MWI What do you mean? If you mean storing the fact that a phone has a message waiting in the database, I don't believe that happens. The RealTime is still in development so it may be added. Matthew - Original Message - From: "Karl H. Putz" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, October 18, 2004 10:00 AM Subject: [Asterisk-Users] New Realtime config and MWI > Does the new Realtime config in the CVS head support setting and clearing > MWI for sip clients? > > Thanks, > > Karl Putz > Forte Communications > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Realtime config and MWI
Does the new Realtime config in the CVS head support setting and clearing MWI for sip clients? Thanks, Karl Putz Forte Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
