[asterisk-users] Req-Installation process for app_dtmftotext.c

[rajesh koniki]

Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried 
with 'spandsp' application for this. But I am getting errors.[I followed the 
instructions at http://www.soft-switch.org/installing-spandsp.html]

specifically by running this command:patch I need clarification on 'ld.so.conf' file.[It has to be in the /etc/ 
directory. If you do not have such file - make one. In the file you need to 
add the path to the spandsp library.] Please give me the steps for this 
step.


I installed asterisk 1.2.17 only, i not installed any libpri or zaptel 
sources.


Can anybody be of help Me on this getting DTMFToText() application on 
asterisk with the help of app_dtmftotext.c and/or spandsp application is 
appreciated.


Regards
K.Rajesh.

_
Spice up your IM conversations. New, colorful and animated emoticons. Get 
chatting! http://server1.msn.co.in/SP05/emoticons/


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Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?

[Nick Seraphin]

On Tue, 15 May 2007, Vincent Delporte wrote:

> Hello,
> 
> In case there are other users of the AsteriskWin32 port...
> 
> I haven't really used the AGI feature of Asterisk to run an application 
> from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, 
> it's also possible to write AGI applications as EXE's (there's a 
> eagi-test.exe file installed by default).
> 
> => When a call comes in, I'd like an AGI application to send an e-mail and 
> send CID name/number to a script on a web server.
> 
> Is this the correct way to do it in Perl, with the modules available in 
> AsteriskWin32? Could I rewrite this in Delphi instead?


ALL AGI scripts are basically just programs that read from stdin and write
to stdout.  They can therefore be written in almost any language.  So yes,
Delphi should work fine.

(I have very fond memories of Delphi, and before that, Borland Pascal w/
Objects for DOS, and before that, Turbo Pascal...  one of these days I'll
have to get the latest version of Delphi and take a walk down memory lane.
These days everything is C this or Perl that.  I loved Pascal. :-)) 

-- Nick


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Re: [asterisk-users] queue_exec: Unable to join queue

[Per Jessen]

lenz wrote:

> Is the queue "enidan" configured at all in queues.conf? and how is it
> defined?
> l.

Sorry, I should have added that:

from queues.conf:

[enidan]
strategy = ringall
;announce = enidan-queue
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]

Also, what I discovered yesterday is the following:

just after an asterisk restart:
*CLI> show queue
enidan   has 0 calls (max unlimited) in 'ringall' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
   No Callers

The "(Invalid)" bit is worrying, but after a reload of app_queue:

*CLI> show queue
enidan   has 0 calls (max unlimited) in 'ringall' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
   No Callers



/Per Jessen, Zürich

-- 
ENIDAN Technologies GmbH - managed email security. 
Starting at SFr1/month/user - http://www.spamchek.ch/

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Re: [asterisk-users] The purpose of DUNDi

[Remco Post]

dave cantera wrote:
> remco, et al,
> could I use dundi where I could use an area code to determine the
> connecting server or dial string?  just like we would use 88XXX to dial
> a 3 digit extension on another server at location 88?  or dial 84XXX for
> a 3 digit extension on a server located at 84?...

yes you can.

You'll setup a context in your dialplan on your server where you'll tell
dundi that you accept calls for say _88XXX and have a mapping for that
context in your dundi.conf

> thanks,
> daveC
> 
> 
> Remco Post wrote:
>> Rilawich Ango wrote:
>>  
>>> It is quite interesting and I am looking for it.  Could you give me
>>> some more information or website how to set it up?
>>>
>>> 
>>
>>
>> Have a look at:
>>
>> http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf
>>
>>
>> and the two links at:
>>
>> http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration
>>
>>
>>   
> 


-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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Re: [asterisk-users] Re: TC400B load problem

[Arun Kumar]

thanks Matthew, I'll try to call Digium.

On 5/14/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:



On May 14, 2007, at 4:53 AM, Arun Kumar wrote:
> Im trying to install my TC400B trans coder card when I do:
>
> modprobe wctc4xxp
>
> tail -f /var/log/messages says:
>
> May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder'
> with 92 transcoders (srcs=000c, dsts=0101)
> May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder'
> with 92 transcoders (srcs=0101, dsts=000c)
> May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps)
> Transcoder support LOADED (firm ver = 56)
> May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed
> with error -5

That looks like a problem that you should talk with Digium Support
about.

Matthew Fredrickson

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Re: [asterisk-users] Simultaneous Capacity

[Armin Schindler]

On Mon, 14 May 2007, Kapil Dhawan wrote:
> Just a quick brief
> 
> I have a requirement of running 10 PRI's (300 Channels). I still have to
> decide on hardware and cards. Can you suggest some. As per my understanding it
> will be tough to go beyond 150.

I didn't test exactly this yet, but from my experience it should work with
the Dialogic DIVA Server 2 x 4PRI + 1 x 2PRI cards.

Armin
 
> Alex Balashov wrote:
> > On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
> > 
> > > I want to try Asterisk with 10 PRI on a single Xeon machine with
> > > g711. Is it feasible.
> > 
> > In truth, it is very unlikely.
> > 
> >  How are you planning to pick up the PRIs, anyway?  3 quad-span T1 cards?
> > 
> > -- 
> > Alex Balashov   <[EMAIL PROTECTED]>
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[asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?

[Vincent Delporte]

Hello,

In case there are other users of the AsteriskWin32 port...

I haven't really used the AGI feature of Asterisk to run an application 
from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, 
it's also possible to write AGI applications as EXE's (there's a 
eagi-test.exe file installed by default).


=> When a call comes in, I'd like an AGI application to send an e-mail and 
send CID name/number to a script on a web server.


Is this the correct way to do it in Perl, with the modules available in 
AsteriskWin32? Could I rewrite this in Delphi instead?


---
#!/usr/bin/perl

;---
;Note: Not sure if *Win32 supports LWP::Simple and Net::SMTP!

;Called from extensions.conf
;exten => group,n,AGI(notify.agi|${CALLERID(num)}|${CALLERID(name)})

;---

use strict;

open STDOUT, '>/dev/null';
fork and exit;

;---
use LWP::Simple;

my $cidnum = $ARGV[0];
my $cidname = $ARGV[1];

my $url = 'http://www.acme.com/input.php?name=$cidname&number=$cidnum';
my $content = get $url;
die "Couldn't get $url" unless defined $content;
print STDERR "Notified web server"

;---
use Net::SMTP;

$smtp = Net::SMTP->new('smtp.acme.com'); # connect to an SMTP server
$smtp->mail( '[EMAIL PROTECTED]' ); # use the sender's address here
$smtp->to('[EMAIL PROTECTED]');# recipient's address
$smtp->data();  # Start the mail

# Send the header.
$smtp->datasend("To: [EMAIL PROTECTED]");
$smtp->datasend("From: [EMAIL PROTECTED]");
$smtp->datasend("\n");

# Send the body.
$smtp->datasend("Call received from $cidname/$cidnum\n");
$smtp->dataend();   # Finish sending the mail
$smtp->quit;
print STDERR "Send e-mail"
---

Thanks for any tip 


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[asterisk-users] How to write data to astdb?

[Vincent Delporte]

Hello,

	I'm trying to fill CID data into the astdb using AsteriskWin32's 
asterisk.exe, to no avail: The batch file stops after the first line, and 
just waits:



rem c:\cygroot\mystuff>import.bat
rem
rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put 
cidname 123 "My cellphone"'

rem
rem Asterisk module loaded successfully
rem Asterisk entry point foundW2003*CLI> Updated database successfully
rem Verbosity is at least 1
rem STUCK HERE!

C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"'
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 456 "This is a test"'


I don't know why the batch script stops after the first line.

So, I installed ActivePerl and the asterisk-perl package from CPAN, and 
tried this, but it doesn't work either:



use Asterisk::AGI;

$AGI = new Asterisk::AGI;
$AGI->database_put('cidname', '', 'my number');


Is there a way to access astdb directly, instead of through an AGI script?

Thank you.

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Re: [asterisk-users] Web based call control

[Nick Seraphin]

On Mon, 14 May 2007, Jordan Novak wrote:

> Does anyone know if it is possible to use a manager command to answer
> an incoming call and not consider it answered unitl it is received.
> Here is an example, I am deivering a call in the dialplan to a home
> telephone number. I don't want his voicemail to answer and I have no
> idea how long it will take to go to their home phone voicemail, but I
> don't want to deliver the call there, I want it to go to the next
> priority in asterisk. So I was thinking that it would be nice to build
> a web interface that they could have a button to answer with. This
> would send a manager command to the server telling it to answer the
> channel, any thoughts on how to do this.
>  


Most companies that I've seen who want this type of behavior have 2
available options.

1) Ask the person with the destination home phone number to cancel their
voicemail from the phone company, so that the phone just rings until it
times out.  Usually they won't want to do this, because other incoming
calls to their home number would not go to voicemail then... so unless
it's a line dedicated as a home-office number that only receives calls
from your system, they won't be happy.

2) The most popular option is have the caller listen to music while
waiting, and then when the destination picks up, play an announcement to
them and wait for DTMF input.  "Hi, you have a call from the PBX system...
press 1 to accept this call."  If they press 1, you connect the two
parties.  If after X number of seconds you don't get a response, i.e.
voicemail picked up, or their 4 yr old child picked up, then no DTMF will
be received and you set it to "time out" and either go on to the next
priority or send it to voicemail.

As far as I know, there is absolutely no way to prevent the destination's
phone company voicemail (or answering machine) from answering the call
unless you have a reliable way to know exactly how many rings it is set to
before it will answer, AND you have assurance that the number of rings 
will NOT be changed.

Granted, probably 90% or more of all voicemail systems out there default
to answer after the 4th or 5th ring, you could always set it to time-out
after 3 rings and have decent success rates... but if the caller is
already on the phone, unless they have call waiting, the call will go to
voicemail on the first ring.

I guess theoretically you could always use voicemail detection to see if
voicemail answered the call instead of a human, and then somehow grab the
call back and transfer it to another extension... but this would probably
require a custom Application or modification of the Asterisk source code.

If I'm wrong, and there's a better way to do this, someone PLEASE let me
know... because I'm planning a project that will have the exact same
problem.  (That's why I've researched this problem already.)

-- Nick


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Re: [asterisk-users] OT (semi) E60 problem

[Martin Joseph]

On May 14, 2007, at 12:34 PM, Tim Panton wrote:



On 14 May 2007, at 17:50, Martin Joseph wrote:


Hello again gurus.

I have been using Asterisk with great results going on a couple of  
years now.


My primary box is running asterisk 1.42 built from a tar ball on  
Mac OSX 10.4.9.


I have a very odd issue that I cannot seem to nail down, which is  
related to my Nokia E60 SIP phone.


I use the E60 with very good results (latest firmware) from  
several locations.


Basically it works fine from everywhere EXCEPT when it's on the  
same LAN as my asterisk box.


The SIP config. that I have setup for the LAN connection refers to  
my asterisk box by it's local IP (ie 192.168.1.101).  The external  
configs refer to the asterisk box by it's name (ie sip.domain.com).


It seems like this has something to do with the authentication  
realm?  If I  create a new config on the phone using the LAN  
address, it works, but then when I leave the LAN, it appears to  
register,  but issues a "connection error" when I try to place calls.


To get it working again from outside the LAN,  I can change the  
realm parameter in SIP.conf, and then reload and then change it  
AGAIN back to it's original value and reload, the phone then works  
fine.   At that point, the LAN based config won't work anymore and  
will give me a "connection error".


Any thoughts on this?  Ideas on how to troubleshoot further or  
work around it would be great.


I had a similar problem, and it turned out to be capitalization error.
The font the nokia uses for it's config dialogs is such that the  
caps don't stand out.
This was compounded by the fact that the default input method  
capitalizes the first letter of every line.


I guess this isn't your problem, but if it is I've saved you a  
_long_ tedious hunt!



Yes, thanks for that thought Tim,  although I think they fixed that  
"feature" in the latest firmware update.  It definitely seems to be  
some sort of realm realated issue (I think)...


I am thinking that if I could get my router to properly resolve the  
name (ie sip.domain.com) into the internal number (ie 192.168.1.101)  
then my issue would be resolved.  I even did attempt setting up a  
seperate DNS for this lookup,  but the stupid phone caches DNS info,  
so that when I leave the site, it continues to produce the (now  
wrong) internal IP address.


Ugh.  I think I am going to try playing with the SIP debug settings  
in asterisk and see if that gives me any additional help on figuring  
out what is going on...


Another solution would be to run and entirely seperate asterisk for  
the LAN, which had another realm name  That seems insane though.


Oh well,
Marty

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[asterisk-users] `PATH_MAX' undeclared here (not in a function) in asterisk!

[lizhong zhu]

hello, asteriskers:
I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run 
./configure and menuselect with embedded modules. but running make comes out 
errors:
ranlib libmxml.a
make[3]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect/mxml'
cc -Wall  -o menuselect.o -g -c -D_GNU_SOURCE menuselect.c
cc -Wall  -o menuselect_curses.o -g -c -D_GNU_SOURCE  menuselect_curses.c
cc -Wall  -o strcompat.o -g -c -D_GNU_SOURCE strcompat.c
cc -g -Wall -o menuselect menuselect.o menuselect_curses.o strcompat.o 
mxml/libmxml.a -lncurses
make[2]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect'
make[1]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect'
menuselect/menuselect --check-deps   menuselect.makeopts
   [CC] stereorize.c -> stereorize.o
   [CC] frame.c -> frame.o
   [LD] stereorize.o frame.o -> stereorize
   [CC] streamplayer.c -> streamplayer.o
   [LD] streamplayer.o -> streamplayer
   [CC] aelparse.c -> aelparse.o
In file included from /usr/src/asterisk-1.4.2/include/asterisk.h:32,
 from ael_lex.c:19:
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:23: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:24: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:25: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:26: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:27: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:28: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:29: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:30: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:31: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:32: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:33: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:34: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:35: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:36: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:37: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:38: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:39: `PATH_MAX' undeclared here 
(not in a function)
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:40: `PATH_MAX' undeclared here 
(not in a function)
In file included from ael_lex.c:19:
does anyone know that problem? please give me a help!
thanks!
zhulizhong

   
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[asterisk-users] Web based call control

[Jordan Novak]

Does anyone know if it is possible to use a manager command to answer an 
incoming call and not consider it answered unitl it is received. Here is an 
example, I am deivering a call in the dialplan to a home telephone number. I 
don't want his voicemail to answer and I have no idea how long it will take to 
go to their home phone voicemail, but I don't want to deliver the call there, I 
want it to go to the next priority in asterisk. So I was thinking that it would 
be nice to build a web interface that they could have a button to answer with. 
This would send a manager command to the server telling it to answer the 
channel, any thoughts on how to do this.
 
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Re: [asterisk-users] Asterisk High-Capacity Stability

[Atlanticnynex]

I'm curious what kind of configuration/features/modules you could recommend
for my setup. Can you explain further what you mean by OpenSER to Asterisk?

Thanks Much,

kn0x

On 5/14/07, EdPimentl <[EMAIL PROTECTED]> wrote:


Actually, OpenSER is just the you will need to scale Asterisk.
We have perform a number of OpenSER to Asterisk implementation for 50k
plus users
-E

On 5/14/07, Atlanticnynex <[EMAIL PROTECTED]> wrote:
>
> Thanks for all the input guys.
> This is what I had originally expected.
> Does anyone have any recommendations for other software configurations?
> I've thought about using OpenSER + rtpproxy(or media proxy), but it seems
> that OpenSER is not designed
> to do this sort of thing and would require some tricky hacking(?). I
> guess I'm wondering if their are any other opensource B2BUA-like
> softswitches that would fit what I'm looking for. What are these VoIP
> carriers using?
>
> Thanks,
>
> kn0x
>
>


--
Thanks in advance and best regards,

Ed Pimentel
AgileCO
Founder

Web:   http://AgileCO.net
Mail:   edpimentl[at]gmail.com
Mail2: edpimentl[at]ieee.org
IM: edpimentl [AOL | Jabber | Yahoo | MSN ]
Voip:   edpimentl [SKype | GoogleTalk ]

Mobile Content Marketing/Management/Digital Delivery
http://mobilecentral.ws

Mobile ( Context Aware, AmbientIntelligence, Location ) based Social
Network
http://TagR.mobi (Alpha)

Mobile Payment - P2P Payment
http://agilepay.ws

[S4]Secure Scalable Streaming Storage GridService
http://DatR.ws

Private Label Social Networks
http://GooGaYa.com

Sponsor of P2PSIP  open source [viasip_ng] project
Based on IETF P2PSIP WG
https://sourceforge.net/projects/viasip/
http://groups.google.com/group/viasip_ng
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RE: [asterisk-users] Proper AGI use with MySQL

[lists]

We are not very strong perl programmers and we've been using the Asterisk::AGI 
library. Is there any sample you can point us that shows how to do this?

Thanks

On Mon, May 14, 2007 9:05 pm, Michelle Dupuis <[EMAIL PROTECTED]> said:

> How about forking the process when the AGI launches, and pass the PID back
> to Asterisk in a variable.  When the call ends (caught at the "h"), call
> another AGI script to kill/stop that pid.
> 

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Re: [asterisk-users] Blind Transfer - Who transferred the call?

[Lee Jenkins]

Lee Jenkins wrote:


Hi all,

Is there a way to tell which extension transferred a call in a blind 
transfer?


Sorry if it's a basic question, but I haven't seen an answer. 
${CALLERID(num)} still holds the outside party caller id (which it 
should), but I'd like to the extension number of the extension that 
transferred the call.




Parsing the variable ${BLINDTRANSFER} gives what I need.

// Sample: AChannel = 'SIP/111-0Asswwosee'
Function DialStringFromChannel(const AChannel: string): string;
var
iPOS: Integer;
begin
   iPOS := POS('-', AChannel);
   if (iPOS = 0) then exit;
   result := Copy(AChannel, 1, iPOS -1);
end;


I had thought that there was a specific variable somewhere for the 
extension that transferred a call.  No matter, the above code works like 
a treat.


--

Warm Regards,

Lee



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RE: [asterisk-users] Proper AGI use with MySQL

[Michelle Dupuis]

How about forking the process when the AGI launches, and pass the PID back
to Asterisk in a variable.  When the call ends (caught at the "h"), call
another AGI script to kill/stop that pid. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, May 14, 2007 5:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Cc: Asterisk Discussion
Subject: RE: [asterisk-users] Proper AGI use with MySQL

Sorry, just to make sure this is clear, in #2 below, when I said "We would
like for the AGI script to stay running for the life of the call...", I also
meant after the call is transfered to the customer service queue. This is so
because we need to note that the call ended (update callend = NOW())
regardless of whether the call stayed only in the IVR or the caller spoke
with a customer service agent.

Thanks again

On Mon, May 14, 2007 5:40 pm, [EMAIL PROTECTED] said:

> Hi,
>
> We have a "simple" AGI script that provides some IVR functionality. It 
> connects to a MySQL database in order to create a call record and capture
IVR data.
>
> During the IVR process, we need to store the time the call started, so 
> basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = 
> AGI(agi_uniqueid). As the user selects different options, we update 
> the row to reflect the user's selection. There are a couple of options 
> within the IVR that allows the user to speak with a live customer 
> service rep. So, in those cases, we do a AGI exec to Dial out to the 
> customer service queue and transfer the caller there. In the dialplan, 
> we have extension h, execute DeadAGI which basically looks up the
agi_uniqueid and updates the time the call ended in MySQL (e.g. callend =
NOW()).
>
> All this seems to be working. However, we just don't feel we are doing 
> things properly and reading up on the wiki more about AGI and dialing 
> out, etc, just makes me feel we could be doing things better.
>
> Here are some of the things we think we could be doing better but are not
sure:
>
> 1) Ideally, we would like for the AGI script to know when the call 
> hangs up so that it properly updates "callend" without having to run 
> the DeadAGI command in the h extension.
>
> 2) We would like for the AGI script to stay running for the life of 
> the call and keep in memory all the user's IVR selections until the 
> call is hung up. At which point, we could actually INSERT the row in 
> MySQL with all the data, instead of constantly hitting the database with
updates.
>
> 3) We read on the wiki the following: "If the AGI application dials 
> outward by executing Dial, control over the call returns to the 
> dialplan and the script loses contact with the Asterisk server. The 
> script continues to run in the background by itself and is free to 
> clean up and do post-dial processing." In our IVR, we always exit with 
> -1. So, this statement confused us. Does it mean that when we transfer 
> the call to the queue, we should actually return 0 instead of -1 to
indicate that the AGI is still running? Can anyone explain this further?
>
> 4) When should we close the database handle? Currently, we have it at 
> the end of the AGI script and also as part of the DeadAGI script. 
> However, which one is actually closing it, we don't know.
>
> Comments are extremely welcomed and appreciated.
>
> Thanks
>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Blind Transfer - Who transferred the call?

[Lee Jenkins]

Hi all,

Is there a way to tell which extension transferred a call in a blind 
transfer?


Sorry if it's a basic question, but I haven't seen an answer. 
${CALLERID(num)} still holds the outside party caller id (which it 
should), but I'd like to the extension number of the extension that 
transferred the call.


Any suggestions?

Thank you,

--

Warm Regards,

Lee




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[asterisk-users] Asterisk Now

[Wiley Siler]

Can someone tell me what is included in this distro?

Does it have voicemail, meetme, panel, and IVR?

 

Thanks,

 

Wiley E. Siler
Director of Information Technology
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailto:[EMAIL PROTECTED]  
www.education2020.com   

 

Helping students on a mission. Graduation and beyond.

 

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Re: [asterisk-users] ast_yyerror - Help

[Steve Murphy]

On Mon, 2007-05-14 at 14:52 -0500, Rob Schall wrote:
> Hey all,
> 
> We're starting to see "all circuits are busy" and a few dropped calls.
> When these happen, in the messages log, I see the following error.
> 
> May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
> syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or
> TOK_LP or TOKEN; Input:
> 0?7:
> 
> What causes this?

The ast_expr2 stuff is what gets called when you have $[...] expressions
in your dialplan.

So, it looks like you have have something like this in your dialplan:

$[ ${var} ? ${var2} : ${var3} ]

and ${var3} evaluates to an empty string (or something).

You need to narrow down where exactly in the dialplan you are when this
error happens. If you have no idea, look for $[ in your dialplan, and
study each one to determine a set of candidates. Once you find the right
expression, then you need to determine why var3 is empty, and maybe
insert some code to make sure it's always set to something, or rephrase
the expression to work better in that case.

Best of luck!

murf

-- 
Steve Murphy
Software Developer
Digium


smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Re: dialplan: execute on hangup

[Michael Kamleitner]

thx a lot Tony, I didn't know about using the h-extension (I'm new to
Asterisk)!

this way it works:

...
exten => s,n,Voicemail(${Enter},u)
exten => s,n,AGI(foneboxx.php|${Enter})
exten => h,1,DeadAGI(foneboxx.php|${Enter})

greetings,
michael



On 5/14/07, Tony Mountifield <[EMAIL PROTECTED]> wrote:


In article <[EMAIL PROTECTED]>,
Michael Kamleitner <[EMAIL PROTECTED]> wrote:
>
> thx Tony, but DeadAGI doesn't seem to fit my needs... the way I
understand
> its functioniality (
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI),
DeadAGI
> is ensureing that an executed AGI-script is finished, even if the caller
> hung up _during_ execution.

This is true, but DeadAGI will also work when the channel is already hung
up.

> in my case, I need to execute the AGI-script _after_ the user hung up &
the
> voicemail is recorded.

That's exactly what DeadAGI is for. I use DeadAGI extensively within the
'h'
extension to do post-call processing for my applications.

> another ideas: is there away to tell the Voicemail-command to execute an
> AGI-script when recording is finsihed?

Not without modifying the source code.

Cheers
Tony

--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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--
Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
https://www.xing.com/profile/Michael_Kamleitner
-
m-otion GmbH
Favoritenstr 4-6/III, 1040 Wien
+43 1 205705 / 21 (Fax 99)
-
www.m-otion.com
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[asterisk-users] Re: dialplan: execute on hangup

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Michael Kamleitner <[EMAIL PROTECTED]> wrote:
> 
> thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand
> its functioniality (
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI
> is ensureing that an executed AGI-script is finished, even if the caller
> hung up _during_ execution.

This is true, but DeadAGI will also work when the channel is already hung up.

> in my case, I need to execute the AGI-script _after_ the user hung up & the
> voicemail is recorded.

That's exactly what DeadAGI is for. I use DeadAGI extensively within the 'h'
extension to do post-call processing for my applications.

> another ideas: is there away to tell the Voicemail-command to execute an
> AGI-script when recording is finsihed?

Not without modifying the source code.

Cheers
Tony

-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] Proper AGI use with MySQL

[lists]

Sorry, just to make sure this is clear, in #2 below, when I said "We would like 
for the AGI script to stay running for the life of the call...", I also meant 
after the call is transfered to the customer service queue. This is so because 
we need to note that the call ended (update callend = NOW()) regardless of 
whether the call stayed only in the IVR or the caller spoke with a customer 
service agent.

Thanks again

On Mon, May 14, 2007 5:40 pm, [EMAIL PROTECTED] said:

> Hi,
> 
> We have a "simple" AGI script that provides some IVR functionality. It 
> connects to
> a MySQL database in order to create a call record and capture IVR data.
> 
> During the IVR process, we need to store the time the call started, so 
> basically
> we INSERT a new MySQL row with callstart = NOW(), uniqueid = 
> AGI(agi_uniqueid). As
> the user selects different options, we update the row to reflect the user's
> selection. There are a couple of options within the IVR that allows the user 
> to
> speak with a live customer service rep. So, in those cases, we do a AGI exec 
> to
> Dial out to the customer service queue and transfer the caller there. In the
> dialplan, we have extension h, execute DeadAGI which basically looks up the
> agi_uniqueid and updates the time the call ended in MySQL (e.g. callend = 
> NOW()).
> 
> All this seems to be working. However, we just don't feel we are doing things
> properly and reading up on the wiki more about AGI and dialing out, etc, just
> makes me feel we could be doing things better.
> 
> Here are some of the things we think we could be doing better but are not 
> sure:
> 
> 1) Ideally, we would like for the AGI script to know when the call hangs up so
> that it properly updates "callend" without having to run the DeadAGI command 
> in
> the h extension.
> 
> 2) We would like for the AGI script to stay running for the life of the call 
> and
> keep in memory all the user's IVR selections until the call is hung up. At 
> which
> point, we could actually INSERT the row in MySQL with all the data, instead of
> constantly hitting the database with updates.
> 
> 3) We read on the wiki the following: "If the AGI application dials outward by
> executing Dial, control over the call returns to the dialplan and the script 
> loses
> contact with the Asterisk server. The script continues to run in the 
> background by
> itself and is free to clean up and do post-dial processing." In our IVR, we 
> always
> exit with -1. So, this statement confused us. Does it mean that when we 
> transfer
> the call to the queue, we should actually return 0 instead of -1 to indicate 
> that
> the AGI is still running? Can anyone explain this further?
> 
> 4) When should we close the database handle? Currently, we have it at the end 
> of
> the AGI script and also as part of the DeadAGI script. However, which one is
> actually closing it, we don't know.
> 
> Comments are extremely welcomed and appreciated.
> 
> Thanks
> 
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Proper AGI use with MySQL

[lists]

Hi,

We have a "simple" AGI script that provides some IVR functionality. It connects 
to a MySQL database in order to create a call record and capture IVR data.

During the IVR process, we need to store the time the call started, so 
basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = 
AGI(agi_uniqueid). As the user selects different options, we update the row to 
reflect the user's selection. There are a couple of options within the IVR that 
allows the user to speak with a live customer service rep. So, in those cases, 
we do a AGI exec to Dial out to the customer service queue and transfer the 
caller there. In the dialplan, we have extension h, execute DeadAGI which 
basically looks up the agi_uniqueid and updates the time the call ended in 
MySQL (e.g. callend = NOW()).

All this seems to be working. However, we just don't feel we are doing things 
properly and reading up on the wiki more about AGI and dialing out, etc, just 
makes me feel we could be doing things better.

Here are some of the things we think we could be doing better but are not sure:

1) Ideally, we would like for the AGI script to know when the call hangs up so 
that it properly updates "callend" without having to run the DeadAGI command in 
the h extension.

2) We would like for the AGI script to stay running for the life of the call 
and keep in memory all the user's IVR selections until the call is hung up. At 
which point, we could actually INSERT the row in MySQL with all the data, 
instead of constantly hitting the database with updates.

3) We read on the wiki the following: "If the AGI application dials outward by 
executing Dial, control over the call returns to the dialplan and the script 
loses contact with the Asterisk server. The script continues to run in the 
background by itself and is free to clean up and do post-dial processing." In 
our IVR, we always exit with -1. So, this statement confused us. Does it mean 
that when we transfer the call to the queue, we should actually return 0 
instead of -1 to indicate that the AGI is still running? Can anyone explain 
this further?

4) When should we close the database handle? Currently, we have it at the end 
of the AGI script and also as part of the DeadAGI script. However, which one is 
actually closing it, we don't know.

Comments are extremely welcomed and appreciated.

Thanks

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Re: [asterisk-users] The purpose of DUNDi

[dave cantera]

remco, et al,
could I use dundi where I could use an area code to determine the 
connecting server or dial string?  just like we would use 88XXX to dial 
a 3 digit extension on another server at location 88?  or dial 84XXX for 
a 3 digit extension on a server located at 84?...

thanks,
daveC


Remco Post wrote:

Rilawich Ango wrote:
  

It is quite interesting and I am looking for it.  Could you give me
some more information or website how to set it up?





Have a look at:

http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf

and the two links at:

http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration

  


--
Building Strong Relationships w/ Intelligent Customer Service
--

Interlocking Business Solutions, LLC
856-380-0894 x5000


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Re: [asterisk-users] Asterisk High-Capacity Stability

[EdPimentl]

Actually, OpenSER is just the you will need to scale Asterisk.
We have perform a number of OpenSER to Asterisk implementation for 50k plus
users
-E

On 5/14/07, Atlanticnynex <[EMAIL PROTECTED]> wrote:


Thanks for all the input guys.
This is what I had originally expected.
Does anyone have any recommendations for other software configurations?
I've thought about using OpenSER + rtpproxy(or media proxy), but it seems
that OpenSER is not designed
to do this sort of thing and would require some tricky hacking(?). I guess
I'm wondering if their are any other opensource B2BUA-like softswitches that
would fit what I'm looking for. What are these VoIP carriers using?

Thanks,

kn0x





--
Thanks in advance and best regards,

Ed Pimentel
AgileCO
Founder

Web:   http://AgileCO.net
Mail:   edpimentl[at]gmail.com
Mail2: edpimentl[at]ieee.org
IM: edpimentl [AOL | Jabber | Yahoo | MSN ]
Voip:   edpimentl [SKype | GoogleTalk ]

Mobile Content Marketing/Management/Digital Delivery
http://mobilecentral.ws

Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network
http://TagR.mobi (Alpha)

Mobile Payment - P2P Payment
http://agilepay.ws

[S4]Secure Scalable Streaming Storage GridService
http://DatR.ws

Private Label Social Networks
http://GooGaYa.com

Sponsor of P2PSIP  open source [viasip_ng] project
Based on IETF P2PSIP WG
https://sourceforge.net/projects/viasip/
http://groups.google.com/group/viasip_ng
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Re: [asterisk-users] zaptel huge irq problem

[Matthew Fredrickson]

You didn't even read the thread before replying.  And for what it is 
worth, we at Digium are very anxious to solve any sort of IRQ problems 
that you (or others) might have.


Matthew Fredrickson

On May 14, 2007, at 1:43 PM, Salvatore Giudice wrote:

Try switching to a Sangoma card. You won’t have anymore  IRQ issues 
once you abandon Digium hardware.

 
--
 Salvatore Giudice
 [EMAIL PROTECTED]

 VoIP Security Training, LLC
 http://VoIPSecurityTraining.com

 848 N. Rainbow Blvd. #1676
 Las Vegas, NV 89107
 Phone: (617) 959-7625
 Fax: (214) 279-2906
 
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of François 
Delawarde

Sent: Monday, May 14, 2007 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel huge irq problem
 
Thanks Michael,

 I've already been through all that unfortunately, and I have a SATA 
drive, so no UDMA mode 2 as far as I know. I'm currently trying 
everything again anyway, but i doubt it will work if nothing worked 
the first time.


 Anyone would know of issues with XEN or SMP (or both) kernel? Do dual 
core AMD64 processors have issues?


 François.



 Michael L. Young wrote:
François,
 
I too had a similar problem and found the information on this page 
helpful:

http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
 
What ended up working for me was changing the UDMA to mode 2 for the 
hard

drive.  Once I did that, this card has worked perfectly for me.
 
Michael L. Young
 
 

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of François Delawarde
Sent: Monday, May 14, 2007 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel huge irq problem
 
Hello,
 
I had noticed strange crackling sound on my phone calls going through 
my
zaptel device (TDM400P), so i decided to check on possible timer 
issue,
and found lots of issues on forums concerning the sensibility of 
zaptel

with IRQs, and tried about everything: moving PCI slots, noapic and
acpi=off boot options, play with different kernel options:
iosched/preemption/timer/..., play with BIOS PCI options, change
priorities, PCI latencies, IRQ balance, smp_afinity, 
but impossible to come up with anything correcting that problem.
 
Any idea about this? Is it possible to force the timer to ztdummy (RTC
timer) when you have a zap card plugged in? It's the only thing i 
could

try to make it work.
 
Thanks,
François.
 
Just in case:
 
- Linux 2.6.18 with debian patches and xen enabled, asterisk running 
on

dom0.
 
- Here is my zttest results under a bit of load:
# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062%
99.121094%
99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469%
99.414062% 99.902344%
99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406%
98.449707% 100.00%
 
 
- The card DOES NOT seem to share interrupts (checked also with 
lspci):

# cat /proc/interrupts
   CPU0  CPU1
  1:   1626  0    Phys-irq  i8042
  6:  3  0    Phys-irq  floppy
  8:  0  0    Phys-irq  rtc
  9:  0  0    Phys-irq  acpi
 14: 63  0    Phys-irq  ide0
 16:  1  0    Phys-irq  libata, eth3
 17:    6762583  0    Phys-irq  libata
 18:  13789  0    Phys-irq  libata
 19:   33459690  0    Phys-irq  eth1
 20:   19864325  0    Phys-irq  sky2, eth0
 21:  269250881  0    Phys-irq  wctdm
256:   77735119  0 Dynamic-irq  timer0
257:    3986325  0 Dynamic-irq  resched0
258: 37  0 Dynamic-irq  callfunc0
259:  0    4652748 Dynamic-irq  resched1
260:  0    139 Dynamic-irq  callfunc1
261:  0   28924306 Dynamic-irq  timer1
262:   1021  0 Dynamic-irq  xenbus
263:  0  0 Dynamic-irq  console
NMI:  0  0
LOC:  0  0
ERR:  0
MIS:  0
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--
 _
François Delawarde
Ingeniero de red
Tel: 918.03.92.51
E-mail: [EMAIL PROTECTED]
 _
WIRELESS MUNDI
http://www.wirelessmundi.com/
C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid
28760 TRES CANTOS (Madrid)
Tlf./Fax: (+34) 918 03 92 51
La info

Re: [asterisk-users] queue_exec: Unable to join queue

[lenz]

Is the queue "enidan" configured at all in queues.conf? and how is it  
defined?

l.


In data Mon, 14 May 2007 13:56:25 +0200, Per Jessen <[EMAIL PROTECTED]> ha  
scritto:



I have a queue defined which I use like this:

exten = _X.(reception),n,Ringing
exten = _X.,n,Queue(enidan,t,,,20)
exten = _X.,n,Voicemail(443,u)
exten = _X.,n,Hangup()


When I start asterisk, this queue doesn't work -

-- Executing [EMAIL PROTECTED]:3] Queue("mISDN/3-u0", "enidan|t|||20")
in new stack
[May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable to
join queue 'enidan'
-- Executing [EMAIL PROTECTED]:4] VoiceMail("mISDN/3-u0", "443|u") in
new stack

But all I need to do to fix it is reload app_queue.  Does anyone know
what's going on?


/Per Jessen, Zürich





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Re: [asterisk-users] Asterisk High-Capacity Stability

[Atlanticnynex]

Thanks for all the input guys.
This is what I had originally expected.
Does anyone have any recommendations for other software configurations? I've
thought about using OpenSER + rtpproxy(or media proxy), but it seems that
OpenSER is not designed
to do this sort of thing and would require some tricky hacking(?). I guess
I'm wondering if their are any other opensource B2BUA-like softswitches that
would fit what I'm looking for. What are these VoIP carriers using?

Thanks,

kn0x

On 5/14/07, Daryl Jurbala <[EMAIL PROTECTED]> wrote:



On May 14, 2007, at 1:29 PM, Zoa wrote:

>
> Several people do use it for handling > 50k minutes a day. (I'm one
> of them).
> Yes, you need to know what you are doing, and have a nice design,
> but it is possible.Our code is only slightly altered. (mainly for
> billing purposes).

That's great if you're good enough/have the time to make that
happen.  But when I have issues and call/pay Digium and don't get
timely or meaningful answers, it's doesn't make for a good business
decision to continue using it for that purpose when I can toss in a
Nextone or Sansay and have it "just work".  All the time.  No
babysitting.  Full professional and timely problem resolution from
the vendor, etc, etc, etc.  Don't even get me started on Digium not
being able to get TC400Bs to properly negotiate g.723.1 5.3k when a
client requests 6.3k first (thank god for Cantata).

I guess it all comes down to whether you want things to just work and
be able to have tier 1/2 support capable of actually doing anything
meaningful, or if you want to have the engineering level people
forced to do all the work.  From my standpoint, the smart business
decision is quite clear.

But, as I said, Asterisk is still driving the feature servers, and
works well for it.  As mentioned by someone else previously in the
thread, it makes a great endpoint.

If you're having good success with it, that's fantastic.  I would
hope that you contribute back to the list how you set things up to
make this a possibility.

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[asterisk-users] Junghanns DuoBRI Card HELP !

[Service]

We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error "layer 1 deactivated (F3)"! The card sees no ISDN
device connected to it, neither in NT or TE modes alike. 

We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and that we should try 1.2. 

We tried 1.2 with their driver but ALAS we get even more error messages!

I don't understand how Junghanns puts an unsupported card on the market, but
if anyone has any ideas, I would appreciate it. This card is long overdue
for a client of ours. 

Apostolos


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[asterisk-users] ast_yyerror - Help

[Rob Schall]

Hey all,

We're starting to see "all circuits are busy" and a few dropped calls.
When these happen, in the messages log, I see the following error.

May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or
TOK_LP or TOKEN; Input:
0?7:

What causes this?
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[asterisk-users] Junghanns DuoBRI Card HELP !

[Service]

I really need help. 

We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error "layer 1 deactivated (F3)"!

We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and that we should try 1.2. 

We tried 1.2 with their driver but ALAS we get even more error messages!

I don't understand how Junghanns puts an unsupported card on the market, but
if anyone has any ideas, I would appreciate it. This card is long overdue
for a client of ours. 

Apostolos 


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[asterisk-users] Junghanns DuoBRI Card HELP !

[Service]

I really need help. 

We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error "layer 1 deactivated (F3)"!

We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and that we should try 1.2. 

We tried 1.2 with their driver but ALAS we get even more error messages!

I don't understand how Junghanns puts an unsupported card on the market, but
if anyone has any ideas, I would appreciate it. This card is long overdue
for a client of ours. 

Apostolos 


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[asterisk-users] Some problems with mysql CDR

[Jason Martin]

Hello,

We have finally upgraded to Asterisk 1.4, however we've run into two issues 
that weren't occurring before the upgrade. 

Issue #1: We're an outgoing call center and need to record all calls. We use 
the uniqueid field in the CDR to match with the recording, which we labeled 
with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct 
in the CDR. Here is the manager event for a call:

Event: Cdr
Privilege: call,all
AccountCode: 6384106:MMI-Y:200705081051010077
Source: 00
Destination: 6398714109927773
DestinationContext: outbound
CallerID: 00
Channel: Zap/15-1
DestinationChannel: SIP/teliax-081ed5b0
LastApplication: NoOp
LastData:
StartTime: 2007-05-08 10:51:04
AnswerTime: 2007-05-08 10:51:05
EndTime: 2007-05-08 11:01:56
Duration: 652
BillableSeconds: 651
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: 1178635864.1510
UserField:

And for that record in the database:

'calldate' '2007-05-08 10:51:04'
'clid' '00'
 'src' '00'
'dst' '6398714109927773'
'dcontext' 'outbound'
'channel' 'Zap/15-1'
'dstchannel' 'SIP/teliax-081ed5b0'
'lastapp' 'NoOp'
'lastdata' '',
'duration' 652, 
'billsec' 651, 
'disposition' 'ANSWERED', 
'amaflags' 3, 
'accountcode' '6384106:MMI-Y:200705081051010077', 
'uniqueid' '51010077', 
'userfield' '', 
'MMI_field' 'not found'

Issue #2: When a call is not answered, a record of that call is written to the 
database, but uniqueid is left blank. The next time a call isn't answered, 
Asterisk complains:

cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate 
entry '' for key 1

I haven't found any other information regarding these errors. I am just 
wondering if they are bugs. Any insight would be appreciated!

-- 
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679

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Re: [asterisk-users] OT (semi) E60 problem

[Tim Panton]

On 14 May 2007, at 17:50, Martin Joseph wrote:


Hello again gurus.

I have been using Asterisk with great results going on a couple of  
years now.


My primary box is running asterisk 1.42 built from a tar ball on  
Mac OSX 10.4.9.


I have a very odd issue that I cannot seem to nail down, which is  
related to my Nokia E60 SIP phone.


I use the E60 with very good results (latest firmware) from several  
locations.


Basically it works fine from everywhere EXCEPT when it's on the  
same LAN as my asterisk box.


The SIP config. that I have setup for the LAN connection refers to  
my asterisk box by it's local IP (ie 192.168.1.101).  The external  
configs refer to the asterisk box by it's name (ie sip.domain.com).


It seems like this has something to do with the authentication  
realm?  If I  create a new config on the phone using the LAN  
address, it works, but then when I leave the LAN, it appears to  
register,  but issues a "connection error" when I try to place calls.


To get it working again from outside the LAN,  I can change the  
realm parameter in SIP.conf, and then reload and then change it  
AGAIN back to it's original value and reload, the phone then works  
fine.   At that point, the LAN based config won't work anymore and  
will give me a "connection error".


Any thoughts on this?  Ideas on how to troubleshoot further or work  
around it would be great.


I had a similar problem, and it turned out to be capitalization error.
The font the nokia uses for it's config dialogs is such that the caps  
don't stand out.
This was compounded by the fact that the default input method  
capitalizes the first letter of every line.


I guess this isn't your problem, but if it is I've saved you a _long_  
tedious hunt!



Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection

[Zeeshan Zakaria]

Actually now I am getting so many other weird problems. First of all, choppy
sound on the receiving end on the test server. I don't understand why all of
a sudden voice will go choppy, when bandwidth and Internet upload and
download speeds are good. On the  production server, it registers but won't
work. Sometimes when dialed, caller will still be listening the ringing tone
even after the phone is picked up, and obviously no sound for the called
party or caller because for caller still no one has picked up the phone. And
when trying to call out, server will return error code 488.

I don't know why but on the test server it all works fine, except for the
choppy sound. Production server is exactly the copy of test server with all
the same settings, but nothing works. I checked every single thing I could,
but all the settings are the same.

Maybe I am still missing some small little step. Its a GXP-2000 phones.
Don't know what to do. On port 5060 all goes back to normal, but won't work
if connected to Rogers Wireless Internet.

On 5/14/07, Gerald A <[EMAIL PROTECTED]> wrote:


Hi Zeeshan,

On 5/13/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
>
> I've solved this problem. It was very easy (only if I knew how to do it
> before). I changed the UDP ports, i.e.
>
> 1. In sip.conf, bindport=5070
> 2. In my IP Phone server settings, www.myserver.com:5070
>
> Now it seems to be working good and I hope there'll be no more problem
> with it.


Sorry for not replying earlier; I got your note late, and then when I woke
up had no Internet. Ah, the joys of Rogers.

I'm glad to hear you solved it -- my only concern would be if you now want
to connect "ordinary" 5060 looking phones. I will do a bit of research, I'm
sure Asterisk can bind to more then one port.

Thanks,
Gerald


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--
Zeeshan A Zakaria
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Re: [asterisk-users] Asterisk High-Capacity Stability

[Daryl Jurbala]

On May 14, 2007, at 1:29 PM, Zoa wrote:



Several people do use it for handling > 50k minutes a day. (I'm one  
of them).
Yes, you need to know what you are doing, and have a nice design,  
but it is possible.Our code is only slightly altered. (mainly for  
billing purposes).


That's great if you're good enough/have the time to make that  
happen.  But when I have issues and call/pay Digium and don't get  
timely or meaningful answers, it's doesn't make for a good business  
decision to continue using it for that purpose when I can toss in a  
Nextone or Sansay and have it "just work".  All the time.  No  
babysitting.  Full professional and timely problem resolution from  
the vendor, etc, etc, etc.  Don't even get me started on Digium not  
being able to get TC400Bs to properly negotiate g.723.1 5.3k when a  
client requests 6.3k first (thank god for Cantata).


I guess it all comes down to whether you want things to just work and  
be able to have tier 1/2 support capable of actually doing anything  
meaningful, or if you want to have the engineering level people  
forced to do all the work.  From my standpoint, the smart business  
decision is quite clear.


But, as I said, Asterisk is still driving the feature servers, and  
works well for it.  As mentioned by someone else previously in the  
thread, it makes a great endpoint.


If you're having good success with it, that's fantastic.  I would  
hope that you contribute back to the list how you set things up to  
make this a possibility.


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Re: [asterisk-users] Simultaneous Capacity

[Noah Miller]

I have a requirement of running 10 PRI's (300 Channels). I still have to
decide on hardware and cards. Can you suggest some. As per my
understanding it will be tough to go beyond 150.

Alex Balashov wrote:
> On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
>
>> I want to try Asterisk with 10 PRI on a single Xeon machine with
>> g711. Is it feasible.
>
>   In truth, it is very unlikely.
>
>   How are you planning to pick up the PRIs, anyway?  3 quad-span T1
> cards?


You'll probably want to look into creating a Dundi cluster.  I would
not recommend putting more than two Quad span T1 cards in a single
machine.  Actually, I probably wouldn't put more than one in a single
machine.

If you use 3 Quad PRI cards each one in a different machine and
configure those 3 machines as a Dundi cluster, you should be able to
fulfill your 10PRI / 300 channel requirement.  You might want to
consider even five, six, or more machines.  It would certainly reduce
the impact of a hardware failure.


- Noah
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RE: [asterisk-users] zaptel huge irq problem

[Salvatore Giudice]

Try switching to a Sangoma card. You won’t have anymore  IRQ issues once you
abandon Digium hardware.

 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of François
Delawarde
Sent: Monday, May 14, 2007 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel huge irq problem

 

Thanks Michael,

I've already been through all that unfortunately, and I have a SATA drive,
so no UDMA mode 2 as far as I know. I'm currently trying everything again
anyway, but i doubt it will work if nothing worked the first time.

Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core
AMD64 processors have issues?

François.



Michael L. Young wrote: 

François,
 
I too had a similar problem and found the information on this page helpful:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
 
What ended up working for me was changing the UDMA to mode 2 for the hard
drive.  Once I did that, this card has worked perfectly for me.
 
Michael L. Young
 
  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of François Delawarde
Sent: Monday, May 14, 2007 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel huge irq problem
 
Hello,
 
I had noticed strange crackling sound on my phone calls going through my
zaptel device (TDM400P), so i decided to check on possible timer issue,
and found lots of issues on forums concerning the sensibility of zaptel
with IRQs, and tried about everything: moving PCI slots, noapic and
acpi=off boot options, play with different kernel options:
iosched/preemption/timer/..., play with BIOS PCI options, change
priorities, PCI latencies, IRQ balance, smp_afinity, 
but impossible to come up with anything correcting that problem.
 
Any idea about this? Is it possible to force the timer to ztdummy (RTC
timer) when you have a zap card plugged in? It's the only thing i could
try to make it work.
 
Thanks,
François.
 
Just in case:
 
- Linux 2.6.18 with debian patches and xen enabled, asterisk running on
dom0.
 
- Here is my zttest results under a bit of load:
# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062%
99.121094%
99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469%
99.414062% 99.902344%
99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406%
98.449707% 100.00%
 
 
- The card DOES NOT seem to share interrupts (checked also with lspci):
# cat /proc/interrupts
   CPU0  CPU1
  1:   1626  0Phys-irq  i8042
  6:  3  0Phys-irq  floppy
  8:  0  0Phys-irq  rtc
  9:  0  0Phys-irq  acpi
 14: 63  0Phys-irq  ide0
 16:  1  0Phys-irq  libata, eth3
 17:6762583  0Phys-irq  libata
 18:  13789  0Phys-irq  libata
 19:   33459690  0Phys-irq  eth1
 20:   19864325  0Phys-irq  sky2, eth0
 21:  269250881  0Phys-irq  wctdm
256:   77735119  0 Dynamic-irq  timer0
257:3986325  0 Dynamic-irq  resched0
258: 37  0 Dynamic-irq  callfunc0
259:  04652748 Dynamic-irq  resched1
260:  0139 Dynamic-irq  callfunc1
261:  0   28924306 Dynamic-irq  timer1
262:   1021  0 Dynamic-irq  xenbus
263:  0  0 Dynamic-irq  console
NMI:  0  0
LOC:  0  0
ERR:  0
MIS:  0
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-- 

 _

François Delawarde

Ingeniero de red

Tel: 918.03.92.51

E-mail: [EMAIL PROTECTED]

 _

WIRELESS MUNDI

http://www.wirelessmundi.com/

C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid

28760 TRES CANTOS (Madrid)

Tlf./Fax: (+34) 918 03 92 51

  _  

La información contenida en este mensaje y en sus archivos adjuntos es
CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda
expresamente prohibida la utilización de la misma por cualquier persona
distinta de los destinatarios de esta comunicación

Re: [asterisk-users] Dry Copper Pair

[Noah Miller]

 Stephen i disagree. growing up in new work city i can say its quite
 easy to get away with it in the city. where i live now in new jersey
 (population of around 6) i wouldnt be able to pull that off.
>>> The world is a big place, and I suppose there's room for all kinds.
>>> In these parts, the vigilance is pretty high. The pillars are
>>> padlocked now; they didn't use to be, and the COs are locked down
>>> like Fort Knox.
>>>
>>> Anyway, I know enough more than one person who has landed in the
>>> clink for treating the telco like a personal lab.
>> what exactly was the charge ?
>
> Perhaps something along the lines of "unauthorised tampering with a
> telecomms installation"?

I wasn't going to bother replying to Jon's post, because, well, some
things aren't worth the bother.

But here it is, for the public good.

First, there's section 326 of the Criminal Code of Canada:

> Theft of telecommunication service
>
> 326. (1) Every one commits theft who fraudulently, maliciously, or without 
colour of right,
>
> (a) abstracts, consumes or uses electricity or gas or causes it to be wasted 
or diverted; or
>
> (b) uses any telecommunication facility or obtains any telecommunication 
service.

Then, there's section 334:

> Punishment for theft
>
> 334. Except where otherwise provided by law, every one who commits theft
>
> (a) is guilty of an indictable offence and liable to imprisonment for a term 
not exceeding ten years, where the property stolen is a testamentary instrument or 
the value of what is stolen exceeds five thousand dollars; or
>
> (b) is guilty
>
> (i) of an indictable offence and is liable to imprisonment for a term not 
exceeding two years, or
>
> (ii) of an offence punishable on summary conviction,
>
> where the value of what is stolen does not exceed five thousand dollars.

The person in question was slapped with a $10,000 fine.

Look, these guys take tampering with wire infrastructure seriously.
There's a reason the addresses aren't published, the buildings
non-descript, and the doors locked nine ways to Sunday.


In the neighborhood where I live in Putnam County NY, Verizon recently
posted a sign for a $50,000 reward for information leading to the
arrest of individuals responsible for tampering with their
infrastructure.  Apparently, someone had repeatedly hacked the same
piece of equipment (don't know what it was - they wouldn't say).

I don't know what the criminal codes say, but it is obviously an
offense for which you can be arrested, and Verizon felt it was
important enough to give away a sizeable sum to defend their equipment
and access to their network.


- Noah
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[asterisk-users] IAX2 peer unreachable in one direction - NAT problem?

[Seb Auriol]

The situation is one of my asterisk servers is behind a NAT firewall and one
is not. Both servers have multiple IAX peers. The NAT firewall has port 4569
mapped through to the asterisk server behind. But, the natted server is
almost permanently unreachable from this non-natted server, even though, the
non-natted server is almost permanently _reachable_ from the natted server.
Details are below with iax2 debug and core debug 3. I actually have an
Asterisk 1.2 and an Asterisk 1.4 server in the non-natted role, and both
have the same issue. However, I have another non-natted server (on a
different ISP) that can talk fine to the natted server.

(IP addresses replaced with names.)

myNonNattedServer*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status

myNattedServUN   myNattedServer  (S)  255.255.255.255  4569 (T)
UNREACHABLE

[May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1154 update_max_nontrunk: New max
nontrunk callno is 7
[May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1252 find_callno: Creating new
call structure 6
[May 14 19:06:05] DEBUG[5551]: chan_iax2.c:1644 send_packet: Sending 12 on
6/0 to myNattedServer:4569
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE

   Timestamp: 00012ms  SCall: 6  DCall: 0 [myNattedServer:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG

   Timestamp: 00012ms  SCall: 5  DCall: 6 [myNattedServer:37657]
Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL

   Timestamp: 0ms  SCall: 6  DCall: 5 [myNattedServer:37657]
[May 14 19:06:05] DEBUG[5546]: chan_iax2.c:4788 raw_hangup: Raw Hangup
myNattedServer:37657, src=6, dst=5
[May 14 19:06:06] DEBUG[5540]: chan_iax2.c:1644 send_packet: Sending 12 on
6/0 to myNattedServer:4569
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE

   Timestamp: 00012ms  SCall: 6  DCall: 0 [myNattedServer:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG

   Timestamp: 00012ms  SCall: 6  DCall: 6 [myNattedServer:37657]
Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL

   Timestamp: 0ms  SCall: 6  DCall: 6 [myNattedServer:37657]
[May 14 19:06:06] DEBUG[5542]: chan_iax2.c:4788 raw_hangup: Raw Hangup
myNattedServer:37657, src=6, dst=6

myNattedServer*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status
myNonNattedSeUN  myNonNattedServ (S)  255.255.255.255  4569 (T)  OK (14
ms)

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE
   Timestamp: 00016ms  SCall: 00010  DCall: 0 [myNonNattedServ:4569]
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1007 update_max_nontrunk: New max
nontrunk callno is 12
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1112 find_callno: Creating new call
structure 11
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet
0, (6, 30)
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 30
received
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6857 socket_read: For call=11, set
last=16
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1515 send_packet: Sending 16 on
11/10 to myNonNattedServ:4569
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG
   Timestamp: 00016ms  SCall: 00011  DCall: 00010 [myNonNattedServ:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL
   Timestamp: 0ms  SCall: 00010  DCall: 00011 [myNonNattedServ:4569]
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet
0, (6, 10)
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 10
received
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7510 socket_read: Immediately
destroying 11, having received INVAL
May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7513 socket_read: Destroying call
11

Also when calls are placed to myNonNattedServer from myNattedServer (which
does work), the channel name is IAX2/myNattedServer:37657-callno, as opposed
to IAX2/myNattedServUserName-53.

(BTW, if I turn off qualify on myNonNattedServer, I can still not make calls
from myNonNattedServer to myNattedServer.)

Any idea what is wrong? This used to work fine (possibly when myNattedServer
was only trying to talk to one asterisk server through the NAT - now it has
3, only one of which is working properly).

Many thanks,

Sebastian

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Re: [asterisk-users] Asterisk High-Capacity Stability

[Matthew J. Roth]

Daryl Jurbala wrote:

There is some light IVR type usage for reporting account balances and 
the like.  With anything more than 80 or 90 calls on the box, the IVR 
prompts start to break up.  Ben through replacing hardware, more 
memory, different Asterisk builds, etc.

Zoa wrote:
Several people do use it for handling > 50k minutes a day. (I'm one of 
them).
Yes, you need to know what you are doing, and have a nice design, but 
it is possible.Our code is only slightly altered. (mainly for billing 
purposes).

Zoa,

I've been experiencing the IVR prompts breaking up, as Daryl mentioned.  
The problem also affects queue announcements, but native music-on-hold 
sounds good.


Have you experienced this problem, and if so what steps did you take to 
correct it?


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Asterisk and unicall + mfcr2 signalling

[Moises Silva]

try using testcall with 255 as debug level and report back results in
order to be able to help you.

http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf


On 5/14/07, Joca Loco <[EMAIL PROTECTED]> wrote:

Hi,

I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I
have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I
can make calls over E1, but can't receive.

Here the CLI when I make a call:

-- Executing [EMAIL PROTECTED]:1] Dial("SIP/23-081cbc40",
"Unicall/g1/91642208|50") in new stack
-- Called g1/91642208
[May 14 10:34:59] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Dialing
[May 14 10:34:59] NOTICE[4620]: chan_unicall.c:1959 unicall_exception:
Exception on 8, channel 1
[May 14 10:35:14] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Alerting
-- Hungup 'UniCall/1-1'
  == Spawn extension (ps5, 006191642208, 1) exited non-zero on
'SIP/23-081cbc40'
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK
[May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Drop call
[May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Release call
-- Unicall/1 released


And here the CLI when I receive a call:

[May 14 10:35:51] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/8 event Detected
[May 14 10:35:52] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/8 event Protocol failure
[May 14 10:35:52] ERROR[2914]: chan_unicall.c:2603 handle_uc_event:
Unicall/8 protocol error. Cause 32772

Any idea why I can't receive calls, and fot Unicall protocol error Cause
32772?

Thanks, Joca Loco.

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--
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Re: [asterisk-users] zaptel huge irq problem

[Per Jessen]

Stephen Bosch wrote:

>> # cat /proc/interrupts
>>   CPU0  CPU1
>>  1:   1626  0Phys-irq  i8042
>>  6:  3  0Phys-irq  floppy
>>  8:  0  0Phys-irq  rtc
>>  9:  0  0Phys-irq  acpi
>> 14: 63  0Phys-irq  ide0
>> 16:  1  0Phys-irq  libata, eth3
>> 17:6762583  0Phys-irq  libata
>> 18:  13789  0Phys-irq  libata
>> 19:   33459690  0Phys-irq  eth1
>> 20:   19864325  0Phys-irq  sky2, eth0
>> 21:  269250881  0Phys-irq  wctdm
>> 256:   77735119  0 Dynamic-irq  timer0
>> 257:3986325  0 Dynamic-irq  resched0
>> 258: 37  0 Dynamic-irq  callfunc0
>> 259:  04652748 Dynamic-irq  resched1
>> 260:  0139 Dynamic-irq  callfunc1
>> 261:  0   28924306 Dynamic-irq  timer1
>> 262:   1021  0 Dynamic-irq  xenbus
> 
> I've never seen cat /proc/interrupts output that looks like that...
> 
> waaaitaminute...
> 
> are you running this in a virtual machine? Or on a machine running
> virtual machines?

It looks like a XEN machine.  Well spotted, Stephen.


/Per Jessen, Zürich

-- 
ENIDAN Technologies GmbH - managed email security. 
Starting at SFr1/month/user - http://www.spamchek.ch/

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[asterisk-users] Areski CDR

[Diego Quintana Cruz]

Hi folks,
I was wondering what happened to Areski CDR viewer that came before
with Freepbx. I've noticed that the live-CD contains Areski but the
repositories don't have it. Will you fix that? or shall I install
Areski from sources?

Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
http://routerman.blogsome.com
http://gst.telecom.pucp.edu.pe
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Re: [asterisk-users] How to bring MoH volume down

[Stephen Bosch]

Alex Balashov wrote:
> On Mon, 14 May 2007, Stephen Bosch said something to this effect:
> 
>> Is there a way to do it for voice mail messages? I have a user who has
>> trouble hearing the voice messages, saying they are too quiet.
> 
>   From a cursory glance at the voicemail settings, I can't see a way.
> The voicemail messages are stored in a fairly coherent directory
> structure and it may be possible to develop some sort of process that
> funnels the
> voicemail recordings through a volume-boosting 'sox' transformation or
> similar, specifically for that user, but other than that I can't think of
> anything off the top of my head.
> 
>   Volume issues tend to have to do with the phone the user is using as
> often as not.  The user might benefit from a phone with decent volume
> controls/output gain boosting.

Well, we've already cranked the hell out of his phone. He says that
phone conversations are at a reasonable volume now, but it's the
voicemail that's a problem. I guess we'd have to tweak the gain on the
TDM card.

(The user is in a rock band. :) )

-Stephen-
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Re: [asterisk-users] zaptel huge irq problem

[Stephen Bosch]

François Delawarde wrote:
> Thanks Michael,
> 
> I've already been through all that unfortunately, and I have a SATA
> drive, so no UDMA mode 2 as far as I know. I'm currently trying
> everything again anyway, but i doubt it will work if nothing worked the
> first time.
> 
> Anyone would know of issues with XEN or SMP (or both) kernel? Do dual
> core AMD64 processors have issues?

Uh, yeah...

Xen has many, many problems with interrupt handling and is utterly
unsuitable for running anything that depends on hardware peripherals. I
speak from very painful experience.

There is no way, under any circumstance, that I would try to run
Asterisk with interface cards in a Xen environment. It's too bad you
wasted so much time trying to fix it -- it's never going to work.

Try ripping Xen out and doing it directly on the physical server. I
think you'll find your problems will go away.

-Stephen-
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Re: [asterisk-users] DTMF not recognizing *

[Alex Balashov]

On Mon, 14 May 2007, Rob Schall said something to this effect:


The problem is with having a "send to voicemail" option. Right now, a
user can press "*5053" and they will be sent directly to that user's
voicemail box, rather than their phone. But when you press "2*5053", it
appears the * is ignored or not sent. I need to find a way to make this
option work.


  You would need some way of debugging what DTMF signals actually appear
in the context of the bearing channels on the E&M trunk.

  Is there any apparent difference with the duration of the tones?  That
is to say, how long one holds down the 2 or the *?  Also, does it vary
depending on the delay?  Is it possible to press 2, wait 3-4 seconds,
and then press * and see it work?

-- Alex

--
Alex Balashov   <[EMAIL PROTECTED]>
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Re: [asterisk-users] Double DTMF digits

[Greg Oliver]

On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote:
> I am actually getting DTMF over SIP when people call in to a clients system 
> that is running a2billing. They are using RFC2833.
> 

If you are using a Cisco router anywhere in the loop, there is a known
bug that causes rfc2833 and inband signalling to cause double DTMF.  It
is fixed in IOS > 12.4.11T

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Re: [asterisk-users] Dry Copper Pair

[Paul]

Stephen Bosch wrote:

>Per Jessen wrote:
>  
>
>>Jon Pounder wrote:
>>
>>
>>
>>>Quoting Stephen Bosch <[EMAIL PROTECTED]>:
>>>
>>>  
>>>
C F wrote:


>Stephen i disagree. growing up in new work city i can say its quite
>easy to get away with it in the city. where i live now in new jersey
>(population of around 6) i wouldnt be able to pull that off.
>  
>
The world is a big place, and I suppose there's room for all kinds.
In these parts, the vigilance is pretty high. The pillars are
padlocked now; they didn't use to be, and the COs are locked down
like Fort Knox.

Anyway, I know enough more than one person who has landed in the
clink for treating the telco like a personal lab.


>>>what exactly was the charge ?
>>>  
>>>
>>Perhaps something along the lines of "unauthorised tampering with a
>>telecomms installation"? 
>>
>>
>
>I wasn't going to bother replying to Jon's post, because, well, some
>things aren't worth the bother.
>
>But here it is, for the public good.
>
>First, there's section 326 of the Criminal Code of Canada:
>
>  
>
>>Theft of telecommunication service
>>
>>326. (1) Every one commits theft who fraudulently, maliciously, or without 
>>colour of right,
>>
>>(a) abstracts, consumes or uses electricity or gas or causes it to be wasted 
>>or diverted; or
>>
>>(b) uses any telecommunication facility or obtains any telecommunication 
>>service.
>>
>>
>
>Then, there's section 334:
>
>  
>
>>Punishment for theft
>>
>>334. Except where otherwise provided by law, every one who commits theft
>>
>>(a) is guilty of an indictable offence and liable to imprisonment for a term 
>>not exceeding ten years, where the property stolen is a testamentary 
>>instrument or the value of what is stolen exceeds five thousand dollars; or
>>
>>(b) is guilty
>>
>>(i) of an indictable offence and is liable to imprisonment for a term not 
>>exceeding two years, or
>>
>>(ii) of an offence punishable on summary conviction,
>>
>>where the value of what is stolen does not exceed five thousand dollars.
>>
>>
>
>The person in question was slapped with a $10,000 fine.
>
>Look, these guys take tampering with wire infrastructure seriously.
>There's a reason the addresses aren't published, the buildings
>non-descript, and the doors locked nine ways to Sunday.
>
>  
>
I will add that utility lines usually have easements for the public and
private land they run across. I signed easements for the overhead power
line, the buried telco cable and the wiring pedestal. They run about 650
feet on my private driveway. They are on my property but climbing the
poles, excavating near the cable or opening the pedestal are forms of
trespass.

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Re: [asterisk-users] 'Invalid characters in name' with asterisk-gui

[bkruse]

This belongs in the asterisk-gui mailing list.

However, I will see what I can do.

-bkruse

FYI. It is just a javascript pattern matching function, its super easy 
to change.




Tom Lobato wrote:


   Hi all!


   Is there a way to asterisk-gui to allow underline (as such cpd_tom) 
in Names? It allows to [di]enable alphanumeric, but not underline 
noway. Why such restriction in asterisk-gui if even asterisk 
users.conf allows (and works fine) it?





   Thank you,


   Tom Lobato
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Re: [asterisk-users] RTP Mixer

[Kapil Dhawan]

Perfect Josh...but if i am running an application which has a capability 
of showing number or participants depending upon CC value, that doesn't 
work. Secondly, Asterisk can show on CLI about current "talking" users 
where it is maintaining talking status but not sending it down the line 
to be used by other apps.


Anyways, i will go with your statement and leave it on core developers 
to comment.


Joshua Colp wrote:

Kapil Dhawan wrote:
I was reading an article on RTP Mixer so started studying about the 
mixing done by Asterisk in MeetMe.  Read that CC should contain the 
no of participants ifupto 15 and CSRC should come, but not getting 
any by asterisk.





I'll just leave it at this so we can all move on with our lives: 
Asterisk isn't totally an RTP Mixer in the sense you are reading 
about. It is an audio mixer. Frame of audio comes in over RTP, gets 
sent in (only the audio portion) to be mixed, frame comes out, gets 
turned into RTP again. The RTP part has no idea that multiple sources 
were mixed together, 'nor should it care. The sources could have been 
Zaptel channels for example in which case they couldn't be added to 
the list.


Joshua Colp
Software Developer
Digium, Inc.
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Re: [asterisk-users] How to bring MoH volume down

[Matt]

Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.



Just one user?  Sounds like a user problem... however, with that said, you
can try increasing your zaptel volumes.
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Re: [asterisk-users] Asterisk High-Capacity Stability

[Zoa]

Several people do use it for handling > 50k minutes a day. (I'm one of 
them).
Yes, you need to know what you are doing, and have a nice design, but it 
is possible.Our code is only slightly altered. (mainly for billing 
purposes).


Zoa

Daryl Jurbala wrote:

On May 12, 2007, at 4:11 PM, Atlanticnynex wrote:


Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since I 
have quite a bit of experience there, and very little with SER. At 
this point, I'm wondering from a dimensioning standpoint, what kind 
of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As 
I said, I don't plan to do any transcoding. I read the voip-info page 
on dimensioning and it seems theres some mixed feelings about 
Asterisk in high-capacity environments. I guess I'm looking for input 
as to whether Asterisk could handle roughly one DS3's worth of calls 
(672 calls) just doing the LCR (I've seen some pre-built LCR apps, 
looks like they all do on-the-fly MySQL queries- I think I'd write my 
own AGI that would use a cache).



With my hardware, could Asterisk run stable for this amount of traffic?
What stability issues does Asterisk have at this scale?



Simply put, NO.  I am on a project now where a client had an OpenSER 
box acting as an SBC and registrar passing traffic to several asterisk 
boxes which are doing LCR lookups on the fly as well as writing custom 
CDRs all through PHP AGI scripts to a Postgres DB.  The Asterisk boxes 
do not scale, and randomly start swallowing calls or, more often, 
restart the process (safe_asterisk is handling this).  There is some 
light IVR type usage for reporting account balances and the like.  
With anything more than 80 or 90 calls on the box, the IVR prompts 
start to break up.  Ben through replacing hardware, more memory, 
different Asterisk builds, etc.


I've had an open issue with Digium support on this for at least a 
couple of weeks, and the best advice so far was "try using the SVN 
build".  That makes things better, but it's still not anywhere close 
to fixed..


It's absolutely incredible that Asterisk works at all for some of the 
situations its been put in - major kudos to the developers.  But I 
don't think using it for what you're talking about is a long-term 
business strategy.  When the highlight of the 1.6 release is bridging 
channels, you know high volume sip to sip usage in a carrier class 
call routing environment is NOT what development is focused on.  And 
that's fine.  If you use a wrench to do the job of a screwdriver, you 
shouldn't complain when you bust your knuckles


That being said, I don't meant to trash Asterisk at all.  It's a 
fantastic feature server, and a great PBX, both of which things I use 
it for very successfully.  I just don't think it's ready to handle 50k 
plus minutes a day SIP to SIP with LCR and billing data, no matter 
what you do with it.  I'm 100% positive there are people out there 
doing it successfully, but those are the exception, not the rule.  And 
I doubt they are running unmodified code.


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Re: [asterisk-users] Simultaneous Capacity

[Kapil Dhawan]

Just a quick brief

I have a requirement of running 10 PRI's (300 Channels). I still have to 
decide on hardware and cards. Can you suggest some. As per my 
understanding it will be tough to go beyond 150.


Alex Balashov wrote:

On Mon, 14 May 2007, Kapil Dhawan said something to this effect:

I want to try Asterisk with 10 PRI on a single Xeon machine with 
g711. Is it feasible.


  In truth, it is very unlikely.

  How are you planning to pick up the PRIs, anyway?  3 quad-span T1 
cards?


--
Alex Balashov   <[EMAIL PROTECTED]>
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Re: [asterisk-users] How to bring MoH volume down

[Alex Balashov]

On Mon, 14 May 2007, Stephen Bosch said something to this effect:

Is there a way to do it for voice mail messages? I have a user who has 
trouble hearing the voice messages, saying they are too quiet.


  From a cursory glance at the voicemail settings, I can't see a way.
The voicemail messages are stored in a fairly coherent directory structure 
and it may be possible to develop some sort of process that funnels the

voicemail recordings through a volume-boosting 'sox' transformation or
similar, specifically for that user, but other than that I can't think of
anything off the top of my head.

  Volume issues tend to have to do with the phone the user is using as
often as not.  The user might benefit from a phone with decent volume
controls/output gain boosting.

-- Alex

--
Alex Balashov   <[EMAIL PROTECTED]>
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[asterisk-users] DTMF not recognizing *

[Rob Schall]

With our current setup, we have an older avaya system which is linked
with our asterisk system via a em wink connection. When you press "2" on
the avaya network, it will jump to our asterisk box and then sends DTMF
digits. Asterisk listens for those numbers and then responses as soon as
it has a match.

The problem is with having a "send to voicemail" option. Right now, a
user can press "*5053" and they will be sent directly to that user's
voicemail box, rather than their phone. But when you press "2*5053", it
appears the * is ignored or not sent. I need to find a way to make this
option work.

Any thoughts?
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Re: [asterisk-users] zaptel huge irq problem

[Stephen Bosch]

Hi, Francois:

François Delawarde wrote:
> Hello,
> 
> I had noticed strange crackling sound on my phone calls going through my
> zaptel device (TDM400P), so i decided to check on possible timer issue,
> and found lots of issues on forums concerning the sensibility of zaptel
> with IRQs, and tried about everything: moving PCI slots, noapic and
> acpi=off boot options, play with different kernel options:
> iosched/preemption/timer/..., play with BIOS PCI options, change
> priorities, PCI latencies, IRQ balance, smp_afinity, 
> but impossible to come up with anything correcting that problem.

What kind of motherboard do you have?

> - The card DOES NOT seem to share interrupts (checked also with lspci):
> # cat /proc/interrupts
>   CPU0  CPU1
>  1:   1626  0Phys-irq  i8042
>  6:  3  0Phys-irq  floppy
>  8:  0  0Phys-irq  rtc
>  9:  0  0Phys-irq  acpi
> 14: 63  0Phys-irq  ide0
> 16:  1  0Phys-irq  libata, eth3
> 17:6762583  0Phys-irq  libata
> 18:  13789  0Phys-irq  libata
> 19:   33459690  0Phys-irq  eth1
> 20:   19864325  0Phys-irq  sky2, eth0
> 21:  269250881  0Phys-irq  wctdm
> 256:   77735119  0 Dynamic-irq  timer0
> 257:3986325  0 Dynamic-irq  resched0
> 258: 37  0 Dynamic-irq  callfunc0
> 259:  04652748 Dynamic-irq  resched1
> 260:  0139 Dynamic-irq  callfunc1
> 261:  0   28924306 Dynamic-irq  timer1
> 262:   1021  0 Dynamic-irq  xenbus

I've never seen cat /proc/interrupts output that looks like that...

waaaitaminute...

are you running this in a virtual machine? Or on a machine running
virtual machines?

-Stephen-
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Re: [asterisk-users] Re: TC400B load problem

[Matthew Fredrickson]

On May 14, 2007, at 4:53 AM, Arun Kumar wrote:

Im trying to install my TC400B trans coder card  when  I do:

modprobe wctc4xxp

tail -f /var/log/messages  says:

May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' 
with 92 transcoders (srcs=000c, dsts=0101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' 
with 92 transcoders (srcs=0101, dsts=000c)
May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) 
Transcoder support LOADED (firm ver = 56)
May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed 
with error -5


That looks like a problem that you should talk with Digium Support 
about.


Matthew Fredrickson

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[asterisk-users] OT (semi) E60 problem

[Martin Joseph]

Hello again gurus.

I have been using Asterisk with great results going on a couple of  
years now.


My primary box is running asterisk 1.42 built from a tar ball on Mac  
OSX 10.4.9.


I have a very odd issue that I cannot seem to nail down, which is  
related to my Nokia E60 SIP phone.


I use the E60 with very good results (latest firmware) from several  
locations.


Basically it works fine from everywhere EXCEPT when it's on the same  
LAN as my asterisk box.


The SIP config. that I have setup for the LAN connection refers to my  
asterisk box by it's local IP (ie 192.168.1.101).  The external  
configs refer to the asterisk box by it's name (ie sip.domain.com).


It seems like this has something to do with the authentication  
realm?  If I  create a new config on the phone using the LAN address,  
it works, but then when I leave the LAN, it appears to register,  but  
issues a "connection error" when I try to place calls.


To get it working again from outside the LAN,  I can change the realm  
parameter in SIP.conf, and then reload and then change it AGAIN back  
to it's original value and reload, the phone then works fine.   At  
that point, the LAN based config won't work anymore and will give me  
a "connection error".


Any thoughts on this?  Ideas on how to troubleshoot further or work  
around it would be great.


Thanks,
Marty
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Re: [asterisk-users] Re: CITEL gateway does it work well?

[Stephen Bosch]

Steven wrote:
> The Citel Handset Gateways were the best option for our scenario.
>  
> The cost per port for the number of buttons on our NEC DTerm/E phones
> was about half.
> Also, no network reengineering.

I've noticed that all the people who have good things to say about them
are using East Asian digital phones (e.g. NEC, Toshiba, Samsung, etc);
the NEC phones I have worked with don't give DTMF feedback in a native
install, so you wouldn't notice much difference.

But the Nortel phones do (or are able to, depending on how the switch is
configured); losing that ability would bother me.

-Stephen-
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Re: [asterisk-users] zaptel huge irq problem

[François Delawarde]

Thanks Michael,

I've already been through all that unfortunately, and I have a SATA 
drive, so no UDMA mode 2 as far as I know. I'm currently trying 
everything again anyway, but i doubt it will work if nothing worked the 
first time.


Anyone would know of issues with XEN or SMP (or both) kernel? Do dual 
core AMD64 processors have issues?


François.



Michael L. Young wrote:

François,

I too had a similar problem and found the information on this page helpful:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting

What ended up working for me was changing the UDMA to mode 2 for the hard
drive.  Once I did that, this card has worked perfectly for me.

Michael L. Young

  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of François Delawarde
Sent: Monday, May 14, 2007 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel huge irq problem

Hello,

I had noticed strange crackling sound on my phone calls going through my
zaptel device (TDM400P), so i decided to check on possible timer issue,
and found lots of issues on forums concerning the sensibility of zaptel
with IRQs, and tried about everything: moving PCI slots, noapic and
acpi=off boot options, play with different kernel options:
iosched/preemption/timer/..., play with BIOS PCI options, change
priorities, PCI latencies, IRQ balance, smp_afinity, 
but impossible to come up with anything correcting that problem.

Any idea about this? Is it possible to force the timer to ztdummy (RTC
timer) when you have a zap card plugged in? It's the only thing i could
try to make it work.

Thanks,
François.

Just in case:

- Linux 2.6.18 with debian patches and xen enabled, asterisk running on
dom0.

- Here is my zttest results under a bit of load:
# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062%
99.121094%
99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469%
99.414062% 99.902344%
99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406%
98.449707% 100.00%


- The card DOES NOT seem to share interrupts (checked also with lspci):
# cat /proc/interrupts
   CPU0  CPU1
  1:   1626  0Phys-irq  i8042
  6:  3  0Phys-irq  floppy
  8:  0  0Phys-irq  rtc
  9:  0  0Phys-irq  acpi
 14: 63  0Phys-irq  ide0
 16:  1  0Phys-irq  libata, eth3
 17:6762583  0Phys-irq  libata
 18:  13789  0Phys-irq  libata
 19:   33459690  0Phys-irq  eth1
 20:   19864325  0Phys-irq  sky2, eth0
 21:  269250881  0Phys-irq  wctdm
256:   77735119  0 Dynamic-irq  timer0
257:3986325  0 Dynamic-irq  resched0
258: 37  0 Dynamic-irq  callfunc0
259:  04652748 Dynamic-irq  resched1
260:  0139 Dynamic-irq  callfunc1
261:  0   28924306 Dynamic-irq  timer1
262:   1021  0 Dynamic-irq  xenbus
263:  0  0 Dynamic-irq  console
NMI:  0  0
LOC:  0  0
ERR:  0
MIS:  0
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--

_

François Delawarde

Ingeniero de red

Tel: 918.03.92.51

E-mail: [EMAIL PROTECTED] 

_

WIRELESS MUNDI

http://www.wirelessmundi.com/

C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid

28760 TRES CANTOS (Madrid)

Tlf./Fax: (+34) 918 03 92 51



La información contenida en este mensaje y en sus archivos adjuntos es 
CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda 
expresamente prohibida la utilización de la misma por cualquier persona 
distinta de los destinatarios de esta comunicación. Si usted ha recibido 
este mensaje por error le rogamos que lo comunique inmediatamente a 
WIRELESS MUNDI y lo borre al igual que todos sus documentos adjuntos. El 
correo electrónico no puede asegurar la confidencialidad ni la 
integridad de sus mensajes por lo que WIRELESS MUNDI no se hace 
responsable de tales errores u omisiones.


--0--

All information in this message and its attachments is confidential and 
may be legally privileged. Only intended recipients are authorized to 

Re: [asterisk-users] How to bring MoH volume down

[Stephen Bosch]

Alex Balashov wrote:
> 
> Zeeshan,
> 
> On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect:
> 
>> MoH volume is uncomfortably high and I want to bring it down. Its
>> mpg123. How can I do it?
> 
>   There are some settings in musiconhold.conf that may yield the desired
> effect:
> 
> [default]
> mode=mp3
> directory=/var/lib/asterisk/moh
> 
> ; valid mode options:
> ; quietmp3  -- default
> ; mp3   -- loud
> ; mp3nb -- unbuffered
> ; quietmp3nb-- quiet unbuffered
> ; custom-- run a custom application
> ; files -- read files from a directory in any Asterisk supported
> format
> 
>   If not, it may come down to adjusting the base amplitude of the entire
> track down.  I don't think there's a way to modify the gain specifically
> for MOH.

Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.

-Stephen-

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Re: [asterisk-users] RTP Mixer

[Joshua Colp]

Kapil Dhawan wrote:
I was reading an article on RTP Mixer so started studying about the 
mixing done by Asterisk in MeetMe.  Read that CC should contain the no 
of participants ifupto 15 and CSRC should come, but not getting any by 
asterisk.





I'll just leave it at this so we can all move on with our lives: 
Asterisk isn't totally an RTP Mixer in the sense you are reading about. 
It is an audio mixer. Frame of audio comes in over RTP, gets sent in 
(only the audio portion) to be mixed, frame comes out, gets turned into 
RTP again. The RTP part has no idea that multiple sources were mixed 
together, 'nor should it care. The sources could have been Zaptel 
channels for example in which case they couldn't be added to the list.


Joshua Colp
Software Developer
Digium, Inc.
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Re: [asterisk-users] Dry Copper Pair

[Stephen Bosch]

Per Jessen wrote:
> Jon Pounder wrote:
> 
>> Quoting Stephen Bosch <[EMAIL PROTECTED]>:
>>
>>> C F wrote:
 Stephen i disagree. growing up in new work city i can say its quite
 easy to get away with it in the city. where i live now in new jersey
 (population of around 6) i wouldnt be able to pull that off.
>>> The world is a big place, and I suppose there's room for all kinds.
>>> In these parts, the vigilance is pretty high. The pillars are
>>> padlocked now; they didn't use to be, and the COs are locked down
>>> like Fort Knox.
>>>
>>> Anyway, I know enough more than one person who has landed in the
>>> clink for treating the telco like a personal lab.
>> what exactly was the charge ?
> 
> Perhaps something along the lines of "unauthorised tampering with a
> telecomms installation"? 

I wasn't going to bother replying to Jon's post, because, well, some
things aren't worth the bother.

But here it is, for the public good.

First, there's section 326 of the Criminal Code of Canada:

> Theft of telecommunication service
> 
> 326. (1) Every one commits theft who fraudulently, maliciously, or without 
> colour of right,
> 
> (a) abstracts, consumes or uses electricity or gas or causes it to be wasted 
> or diverted; or
> 
> (b) uses any telecommunication facility or obtains any telecommunication 
> service.

Then, there's section 334:

> Punishment for theft
> 
> 334. Except where otherwise provided by law, every one who commits theft
> 
> (a) is guilty of an indictable offence and liable to imprisonment for a term 
> not exceeding ten years, where the property stolen is a testamentary 
> instrument or the value of what is stolen exceeds five thousand dollars; or
> 
> (b) is guilty
> 
> (i) of an indictable offence and is liable to imprisonment for a term not 
> exceeding two years, or
> 
> (ii) of an offence punishable on summary conviction,
> 
> where the value of what is stolen does not exceed five thousand dollars.

The person in question was slapped with a $10,000 fine.

Look, these guys take tampering with wire infrastructure seriously.
There's a reason the addresses aren't published, the buildings
non-descript, and the doors locked nine ways to Sunday.

-Stephen-

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RE: [asterisk-users] Call to Skype network

[Dave Bour]

Here's my instructions...based off Tim Hunt's great script...needs
cleanup but the gist is hear to get someone going...you may think I'm
reboot happy as there's more than a couple here but past experience
found that reloads didn't do it...reboot seem to get things
going...probably something simple...haven't taken time to resolve.

#get gcc, qt-devel, kernel-devel, asterisk-devel if not installed
#check if security patches applied, if kernel-devel and asterisk-devel
done via source...don't yum install those.

yum install gcc qt-devel kernel-devel.i686 asterisk-devel

# download and install the timhunt chanskype installation...run the
installer file

wget http://www.timhunt.net/stuff/chanskype/chanskype-trixbox.tgz

# run makeaccount.sh 1 (or however many accounts you want to make)
./makeaccount.sh 1

#remove twm and install ratpoison per notes from ChanSkype
yum remove twm
wget
http://savannah.nongnu.org/download/ratpoison/ratpoison-1.4.1.tar.gz
tar -zxf rat
cd ratp
./configure
make
make install

# create autostart for vncserver using "ntsysv"
ntsysv
#find vncserver and mark it for autostart.

#reboot
reboot

# download latest release of chanskype
# located at ftp://ftp.chanskype.com/download/packages/
# find the redhat enterprise version (usually FC3), ie ends with
FC3-RHEL4.bin
# ie... chanskype-1.2.9-FC3-RHEL4.bin
wget ftp://ftp.chanskype.com/download/packages/chanskype*FC3-RHEL4.bin
chmod +x chanskype*

./chanskype-x.x.x.-FC3-RHEL4.bin

# where x.x.x is the release you downloaded.
#if it fails and running trixbox repeat with 1DOT2 after the command 
./chanskype-x.x.x.-FC3-RHEL4.bin 1DOT2
 


# access vnc for your server for each port and log into skype with your
account info.
# ie my server is 192.168.101.150:1 (first port assigned for skype1
account)

# move license if exists to proper location
mkdir /var/lib/instant
mkdir /var/lib/instant/licenses
cp ~/CS*  /var/lib/instant/licenses

reboot


edit /etc/asterisk/skype.conf for number of channels in use (same as the
number from makeaccount).
nano /etc/asterisk/skype.conf


open freepbx...create custom trunk...add dial plan as desired...preface
any dial with 00+ ie..

001+nxxnxx
00+1nxxnxx

then under outgoing settings, custom dial string... add:

Local/[EMAIL PROTECTED]

using config editor...add following 3 lines to extensions_custom

[skype]
exten => _X.,1,Dial(Skype/any/${EXTEN})
exten => _X.,2,Hangup

#save settings.
#reboot
reboot

#using vnc ...for each account created, assign user account in skype.
Save settings...

# now the catch that makes Chanskype annoying... Resolve the prompt that
makes skype prompt you to allow chanskype access otherwise you fail to
get a channel assigned.
#cd to your account created for skype...ie skype1...
cd .skype
cd to your login account on skype
edit config.xml and find line "..change
to chan_skype

#reboot for last time...
reboot

#you are ready to go.  Log into your box, and connect to the asterisk
session and check your channels...should show as 1 (or however many you
purchased) licensed channels.

/usr/sbin/asterisk -rvx "skype status"
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Re: [asterisk-users] Simultaneous Capacity

[Alex Balashov]

On Mon, 14 May 2007, Kapil Dhawan said something to this effect:

I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is 
it feasible.


  In truth, it is very unlikely.

  How are you planning to pick up the PRIs, anyway?  3 quad-span T1 cards?

--
Alex Balashov   <[EMAIL PROTECTED]>
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[asterisk-users] Difference between making a call and Originate

[Christopher Robinson]

When I make a regular call from my SIP phone connected to my Asterisk 
server I have no issues, however when I make a call using Originate :

'Channel'=>"SIP/[EMAIL PROTECTED]",
'Context'=>'mycontext',
'Exten'=>'899',
'Priority'=>1,
'Callerid'=>'whatever'));

It creates a screech sound when the first audio file is played.  Doesn't 
seem to happen with another VSP I tried, but still, why would a regular 
outbound call work just fine and Originate create this strange sound.  I 
know for sure that it isn't the audio file that I'm playing by the way.


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Re: [asterisk-users] How obtain the slot position when a call is parked?

[Andrew Kohlsmith]

On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote:
> I want to ask you if asterisk, when I use the command park(), gives me for
> example a variable that contains the slot position where it parks the call
> or if it only tells me (audio) in the channel this position number? In
> other words, is there a way to obtain and use the value of the slot
> position when the call is parked? Thanks.

No.  You need to use ParkAndAnnounce and a feature I'd managed to get added 
which lets you get the parking slot number in the dialplan variable 
${PARKEDAT}.

I use it like this:

exten => _X.,1,...
exten => _X.,n,ParkAndAnnounce(PARKED,,Local/[EMAIL PROTECTED])

...

[parkinginfo]
exten => s,1,NoOp(PARKEDAT=${PARKEDAT})
exten => s,n,...

So basically when the call gets parked, it "announces" the parking slot to a 
Local channel which executes in [parkinginfo].  Parkinginfo can write it to a 
db, SMS it to a skywriter, whatever you want.

It'd be nice to get this sent to a SIP phone and make parking just that much 
more useful, but as the guys say... patches welcome.  :-)

-A.
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[asterisk-users] Simultaneous Capacity

[Kapil Dhawan]

Hi List

I want to try Asterisk with 10 PRI on a single Xeon machine with g711. 
Is it feasible.




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Re: [asterisk-users] RTP Mixer

[Kapil Dhawan]

I was reading an article on RTP Mixer so started studying about the 
mixing done by Asterisk in MeetMe.  Read that CC should contain the no 
of participants ifupto 15 and CSRC should come, but not getting any by 
asterisk.



Tzafrir Cohen wrote:

On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote:
  

Hi

Can somebody brief me the working of RTP mixer from MeetMe perspective.



(RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get 
mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer)


Aparantly people either don't know enough or don't have the time.

Try rephrasing your question so it will be more specific and thus also 
hopefully take shorter time to answer.


Do you have a working system? Do you need to set up one? What version 
of Asterisk? What types of channels do you try to mix?


  





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RE: [asterisk-users] zaptel huge irq problem

[Michael L. Young]

François,

I too had a similar problem and found the information on this page helpful:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting

What ended up working for me was changing the UDMA to mode 2 for the hard
drive.  Once I did that, this card has worked perfectly for me.

Michael L. Young

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of François Delawarde
> Sent: Monday, May 14, 2007 10:24 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] zaptel huge irq problem
> 
> Hello,
> 
> I had noticed strange crackling sound on my phone calls going through my
> zaptel device (TDM400P), so i decided to check on possible timer issue,
> and found lots of issues on forums concerning the sensibility of zaptel
> with IRQs, and tried about everything: moving PCI slots, noapic and
> acpi=off boot options, play with different kernel options:
> iosched/preemption/timer/..., play with BIOS PCI options, change
> priorities, PCI latencies, IRQ balance, smp_afinity, 
> but impossible to come up with anything correcting that problem.
> 
> Any idea about this? Is it possible to force the timer to ztdummy (RTC
> timer) when you have a zap card plugged in? It's the only thing i could
> try to make it work.
> 
> Thanks,
> François.
> 
> Just in case:
> 
> - Linux 2.6.18 with debian patches and xen enabled, asterisk running on
> dom0.
> 
> - Here is my zttest results under a bit of load:
> # ./zttest
> Opened pseudo zap interface, measuring accuracy...
> 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062%
> 99.121094%
> 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469%
> 99.414062% 99.902344%
> 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406%
> 98.449707% 100.00%
> 
> 
> - The card DOES NOT seem to share interrupts (checked also with lspci):
> # cat /proc/interrupts
>CPU0  CPU1
>   1:   1626  0Phys-irq  i8042
>   6:  3  0Phys-irq  floppy
>   8:  0  0Phys-irq  rtc
>   9:  0  0Phys-irq  acpi
>  14: 63  0Phys-irq  ide0
>  16:  1  0Phys-irq  libata, eth3
>  17:6762583  0Phys-irq  libata
>  18:  13789  0Phys-irq  libata
>  19:   33459690  0Phys-irq  eth1
>  20:   19864325  0Phys-irq  sky2, eth0
>  21:  269250881  0Phys-irq  wctdm
> 256:   77735119  0 Dynamic-irq  timer0
> 257:3986325  0 Dynamic-irq  resched0
> 258: 37  0 Dynamic-irq  callfunc0
> 259:  04652748 Dynamic-irq  resched1
> 260:  0139 Dynamic-irq  callfunc1
> 261:  0   28924306 Dynamic-irq  timer1
> 262:   1021  0 Dynamic-irq  xenbus
> 263:  0  0 Dynamic-irq  console
> NMI:  0  0
> LOC:  0  0
> ERR:  0
> MIS:  0
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RE: [asterisk-users] OT ? Number portability, land line to Cell

[Don Kelly]

I think Joe's analysis is unreasonably negative regarding the landline
companies' willingness to port. The link he provides,
http://www.fcc.gov/cgb/consumerfacts/numbport.html, reflects my experience.

A couple cautions, however:

Landline companies may take two to three weeks to actually complete the port
(as the FCC says, DO NOT cancel your current service until the new service
is actually working).

Your new carrier will request an LOA (Letter of Authorization) to complete
the port. Make sure that the LOA is limited to making changes only to the
service that you want them to be changing and the account title (for your
existing service), service address, account number, etc., are exactly
correct on the LOA. Otherwise you'll hear from your new carrier in a couple
weeks that the old carrier refuses to complete the port because the existing
customer is "ABC Enterprises" and the new customer is "A. B. Cooper
Enterprises." (This is why they may request a copy of your existing phone
bill--to make sure everything is letter-perfect.)

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco
Sent: Monday, May 14, 2007 9:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] OT ? Number portability, land line to Cell

> Having had various issues with local vendor (begins with "V"). am looking
to move to all wireless.  Anyone know if current vendor can refuse to port
the current land line numbers to a wireless provider?
> 
> >From what I've read, the Fed's seem to say "no", they cannot refuse, or
impede this.

Your local Vendor can certainly refuse to port the number, regardless of
whether or not they're actually supposed to allow portability.  They're 
the phone company, they don't have to care.

Excuses can range fom "we don't support that" to "the equipment's too old"
to "my dog ate my homework."

They know that 99.9% of all consumers are stupid and/or will not argue the
point.  Most people do not choose to engage big businesses over things 
like this.  That's unfortunate, of course, because it enables companies to
get away with blowoffs like this successfully and makes it harder for the
rest of us to fight.

You might find it interesting and/or useful to see if you can get them to
port it to their own wireless division, assuming that they have one.

If you decide to press the point, which you're encouraged to do, then the
following resource ought to be helpful.

http://www.fcc.gov/cgb/consumerfacts/numbport.html

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then
I
won't contact you again." - Direct Marketing Ass'n position on e-mail
spam(CNN)
With 24 million small businesses in the US alone, that's way too many
apples.
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[asterisk-users] How is Context Determined when Transferring a Call?

[Brent Torrenga]

When trasferring a call, how is the context determined?

When using a zap device, and the DTMF code for blind or attended transfer is
entered, does the tranfer originate at the context the zap device is set to
be in, or does it originate from where the outside call being transferred
originated in, or the context the current call is in?

I ask because I am seeing strange behavior when trying to transfer some
calls placed on zap devices. Dial plan logic does goto's to get the call to
a context, i.e., internal or PSTN contexts. Some transfers fail though,
because, say the call was originally to the PSTN context, and a transfer
fails if it is to an internal context.

Instead of goto's, is it best to use macros? Or a bunch of include
statements?


Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com

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Re: [asterisk-users] Asterisk High-Capacity Stability

[Alex Balashov]

On Mon, 14 May 2007, Daryl Jurbala said something to this effect:

That being said, I don't meant to trash Asterisk at all.  It's a 
fantastic feature server, and a great PBX, both of which things I use it 
for very successfully.


  Agreed.  And, it's worth pointing out, that's what Asterisk is intended 
to be at this point;  it's an *endpoint*, a UA.  Excellent as a feature 
server, voicemail depository, PBX, IVR, what have you, *not* as a router or 
a PC-host based softswitch.  About the only possible use I could imagine 
for such a thing in a routing scenario is as a broker that commands 
superior intelligence and is able to use extensive logic in call decisions 
(like LCR) and then releases itself from the media and signaling path 
entirely, but if you want that, you can't really use a B2BUA, and a SIP 
proxy like OpenSER can do that much better.  Fast.  Because that's what 
it's designed to do.


  I am not sure why so many people want to use it in call routing 
scenarios, because it's not a transit system.  That's what optimised

network elements are for;  media gateways, proxies, etc.

--
Alex Balashov   <[EMAIL PROTECTED]>
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[asterisk-users] ChanSpy

[Asterisk]

Hi Guys,

Does anyone know if is it possible to put one channel in two different
spygroups?

Thanks! Alex

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Re: [asterisk-users] OT ? Number portability, land line to Cell

[Joe Greco]

> Having had various issues with local vendor (begins with "V"). am looking to 
> move to all wireless.  Anyone know if current vendor can refuse to port the 
> current land line numbers to a wireless provider?
> 
> >From what I've read, the Fed's seem to say "no", they cannot refuse, or 
> >impede this.

Your local Vendor can certainly refuse to port the number, regardless of
whether or not they're actually supposed to allow portability.  They're 
the phone company, they don't have to care.

Excuses can range fom "we don't support that" to "the equipment's too old"
to "my dog ate my homework."

They know that 99.9% of all consumers are stupid and/or will not argue the
point.  Most people do not choose to engage big businesses over things 
like this.  That's unfortunate, of course, because it enables companies to
get away with blowoffs like this successfully and makes it harder for the
rest of us to fight.

You might find it interesting and/or useful to see if you can get them to
port it to their own wireless division, assuming that they have one.

If you decide to press the point, which you're encouraged to do, then the
following resource ought to be helpful.

http://www.fcc.gov/cgb/consumerfacts/numbport.html

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [asterisk-users] How to bring MoH volume down

[Matt]

Please provide us with your config in musiconhold.conf so I/we can see how
you are streaming.   There may be a way to lower the volume, but it depends
on how you are performing the streaming.

On 5/14/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:


Here the problem is that it is streaming audio from the Internet and I
can't lower its volume.

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[asterisk-users] How obtain the slot position when a call is parked?

[lavarini]

Hi,
I want to ask you if asterisk, when I use the command park(), gives me for
example a variable that contains the slot position where it parks the call
or if it only tells me (audio) in the channel this position number? In
other words, is there a way to obtain and use the value of the slot
position when the call is parked? Thanks.

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Re: [asterisk-users] Asterisk High-Capacity Stability

[Daryl Jurbala]

On May 12, 2007, at 4:11 PM, Atlanticnynex wrote:


Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since  
I have quite a bit of experience there, and very little with SER.  
At this point, I'm wondering from a dimensioning standpoint, what  
kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB  
RAM). As I said, I don't plan to do any transcoding. I read the  
voip-info page on dimensioning and it seems theres some mixed  
feelings about Asterisk in high-capacity environments. I guess I'm  
looking for input as to whether Asterisk could handle roughly one  
DS3's worth of calls (672 calls) just doing the LCR (I've seen some  
pre-built LCR apps, looks like they all do on-the-fly MySQL  
queries- I think I'd write my own AGI that would use a cache).



With my hardware, could Asterisk run stable for this amount of  
traffic?

What stability issues does Asterisk have at this scale?



Simply put, NO.  I am on a project now where a client had an OpenSER  
box acting as an SBC and registrar passing traffic to several  
asterisk boxes which are doing LCR lookups on the fly as well as  
writing custom CDRs all through PHP AGI scripts to a Postgres DB.   
The Asterisk boxes do not scale, and randomly start swallowing calls  
or, more often, restart the process (safe_asterisk is handling  
this).  There is some light IVR type usage for reporting account  
balances and the like.  With anything more than 80 or 90 calls on the  
box, the IVR prompts start to break up.  Ben through replacing  
hardware, more memory, different Asterisk builds, etc.


I've had an open issue with Digium support on this for at least a  
couple of weeks, and the best advice so far was "try using the SVN  
build".  That makes things better, but it's still not anywhere close  
to fixed..


It's absolutely incredible that Asterisk works at all for some of the  
situations its been put in - major kudos to the developers.  But I  
don't think using it for what you're talking about is a long-term  
business strategy.  When the highlight of the 1.6 release is bridging  
channels, you know high volume sip to sip usage in a carrier class  
call routing environment is NOT what development is focused on.  And  
that's fine.  If you use a wrench to do the job of a screwdriver, you  
shouldn't complain when you bust your knuckles


That being said, I don't meant to trash Asterisk at all.  It's a  
fantastic feature server, and a great PBX, both of which things I use  
it for very successfully.  I just don't think it's ready to handle  
50k plus minutes a day SIP to SIP with LCR and billing data, no  
matter what you do with it.  I'm 100% positive there are people out  
there doing it successfully, but those are the exception, not the  
rule.  And I doubt they are running unmodified code.


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Re: [asterisk-users] OT ? Number portability, land line to Cell

[SIP]

Joe acquisto wrote:

Having had various issues with local vendor (begins with "V"). am looking to 
move to all wireless.  Anyone know if current vendor can refuse to port the current land 
line numbers to a wireless provider?

>From what I've read, the Fed's seem to say "no", they cannot refuse, or impede 
this.

joe a.

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It depends on whether or not your local vendor actually owns the number 
itself or acquires it from another vendor. If they own the number, I 
don't think they're legally allowed to refuse to port it. If they lease 
the number from another provider, they're not actually obligated to 
assist you in porting, although they might, if you ask nicely, tell you 
with whom you need to deal to port the number.

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Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection

[Gerald A]

Hi Zeeshan,

On 5/13/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:


I've solved this problem. It was very easy (only if I knew how to do it
before). I changed the UDP ports, i.e.

1. In sip.conf, bindport=5070
2. In my IP Phone server settings, www.myserver.com:5070

Now it seems to be working good and I hope there'll be no more problem
with it.



Sorry for not replying earlier; I got your note late, and then when I woke
up had no Internet. Ah, the joys of Rogers.

I'm glad to hear you solved it -- my only concern would be if you now want
to connect "ordinary" 5060 looking phones. I will do a bit of research, I'm
sure Asterisk can bind to more then one port.

Thanks,
Gerald
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[asterisk-users] zaptel huge irq problem

[François Delawarde]

Hello,

I had noticed strange crackling sound on my phone calls going through my 
zaptel device (TDM400P), so i decided to check on possible timer issue, 
and found lots of issues on forums concerning the sensibility of zaptel 
with IRQs, and tried about everything: moving PCI slots, noapic and 
acpi=off boot options, play with different kernel options: 
iosched/preemption/timer/..., play with BIOS PCI options, change 
priorities, PCI latencies, IRQ balance, smp_afinity,  
but impossible to come up with anything correcting that problem.


Any idea about this? Is it possible to force the timer to ztdummy (RTC 
timer) when you have a zap card plugged in? It's the only thing i could 
try to make it work.


Thanks,
François.

Just in case:

- Linux 2.6.18 with debian patches and xen enabled, asterisk running on 
dom0.


- Here is my zttest results under a bit of load:
# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094%
99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 
99.414062% 99.902344%
99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 
98.449707% 100.00%



- The card DOES NOT seem to share interrupts (checked also with lspci):
# cat /proc/interrupts
  CPU0  CPU1
 1:   1626  0Phys-irq  i8042
 6:  3  0Phys-irq  floppy
 8:  0  0Phys-irq  rtc
 9:  0  0Phys-irq  acpi
14: 63  0Phys-irq  ide0
16:  1  0Phys-irq  libata, eth3
17:6762583  0Phys-irq  libata
18:  13789  0Phys-irq  libata
19:   33459690  0Phys-irq  eth1
20:   19864325  0Phys-irq  sky2, eth0
21:  269250881  0Phys-irq  wctdm
256:   77735119  0 Dynamic-irq  timer0
257:3986325  0 Dynamic-irq  resched0
258: 37  0 Dynamic-irq  callfunc0
259:  04652748 Dynamic-irq  resched1
260:  0139 Dynamic-irq  callfunc1
261:  0   28924306 Dynamic-irq  timer1
262:   1021  0 Dynamic-irq  xenbus
263:  0  0 Dynamic-irq  console
NMI:  0  0
LOC:  0  0
ERR:  0
MIS:  0
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Re: [asterisk-users] OT ? Number portability, land line to Cell

[Robert A. Rawlinson]

I was going to port a number here in Ohio and Verizon said it would cost 
$90 to do so as they can charge what it cost them.

Bob R
Joe acquisto wrote:

Having had various issues with local vendor (begins with "V"). am looking to 
move to all wireless.  Anyone know if current vendor can refuse to port the current land 
line numbers to a wireless provider?

>From what I've read, the Fed's seem to say "no", they cannot refuse, or impede 
this.

joe a.

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Re: [asterisk-users] OT ? Number portability, land line to Cell

[Jon Pounder]

Quoting Joe acquisto <[EMAIL PROTECTED]>:

Having had various issues with local vendor (begins with "V"). am   
looking to move to all wireless.  Anyone know if current vendor can   
refuse to port the current land line numbers to a wireless provider?


From what I've read, the Fed's seem to say "no", they cannot   
refuse, or impede this.


In Canada I've had limited success - personal lines seem easier to get  
done than business lines for some reason, and the whole issue of the  
overlapped area codes in the toronto area complicate things further  
(really or just portrayed that way as an excuse I am not really sure.)






joe a.

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Jon Pounder

   _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
_/_/_/  _/  _/ _/_/_/  _/  _/_/
   _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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Re: [asterisk-users] OT ? Number portability, land line to Cell

[Alex Balashov]

On Mon, 14 May 2007, Joe acquisto said something to this effect:

Having had various issues with local vendor (begins with "V"). am looking 
to move to all wireless.  Anyone know if current vendor can refuse to 
port the current land line numbers to a wireless provider?


  LNP does provide for this, at least in principle.

--
Alex Balashov   <[EMAIL PROTECTED]>
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Re: [asterisk-users] How to bring MoH volume down

[Zeeshan Zakaria]

Here the problem is that it is streaming audio from the Internet and I can't
lower its volume.
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[asterisk-users] Codename Pineapple - Chan_sip3 - what's the status?

[Olle E Johansson]

Friends,

I have gotten a few questions lately on the status on the Codename  
Pineapple project, the project
that hopefully will produce a more stable and SIP compliant SIP stack  
for Asterisk.


Due to lack of funding, it's postponed until further notice.

I have a few sponsors, but not enough to be able to dedicate time to  
work on it. And since Digium
hasn't made up their minds after thinking about it for more than a  
year, recent changes has not been

updated on svn.digium.com

The work that has been done so far, to mention some major issues

- New configuration parser
- New device type: phone (no more peers/users)
- New way to handle messages (much less copying of in-memory data)
- New transaction engine started
- Adjustable SIP timers
- Split into multiple source code files
- Call pickup support
- New registration handling

This work has been sponsored by Edvina and Voop.

Also, a lot of general cleanup and a new abstraction to prepare   
handling multiple sockets
and domain-level configurations. I've gone through and changed quite  
a lot of the source.
The Codename Pineapple SIP stack is already far away from chan_sip.c,  
but not anywhere
close to something I would start testing. Work in progress or a sad  
ruin... You choose.


I can't resist working on it now and then, but don't expect any major  
progress.


If you have ideas on how to get the community to help fund a major  
overhaul like this,
please send me e-mail off list. To find out more about Codename  
Pineapple, please visit

http://www.codename-pineapple.org

A big Thank You to Voop, Nuvio, TransNexus and Peter Gradwell for  
your support!


Best regards,
/Olle
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[asterisk-users] OT ? Number portability, land line to Cell

[Joe acquisto]

Having had various issues with local vendor (begins with "V"). am looking to 
move to all wireless.  Anyone know if current vendor can refuse to port the 
current land line numbers to a wireless provider?

>From what I've read, the Fed's seem to say "no", they cannot refuse, or impede 
>this.

joe a.

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[asterisk-users] Re: CITEL gateway does it work well?

[Steven]

The Citel Handset Gateways were the best option for our scenario.

The cost per port for the number of buttons on our NEC DTerm/E phones was about 
half.
Also, no network reengineering.

We connected new 66 blocks to the Citel units. And just cutover from the old to 
the new.
When you configure the extension on on the Citel, it does not register to 
asterisk until a phone is connected.

Someday, I will upgrade our network, but using the old phones is much, much 
simpler at this point.

Also, the way they did their BLFs is very asterisk freindly.  

We have been very happy with our units.




-- 
-- 
Steven

http://www.glimasoutheast.org



  "Robert Augustyn" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
  Hi all,
  Is using a Citel gateway with Asterisk a good solution for reusing of the old 
Nortel digital phones?
  Would love to get some input from actual users.
  Any/all opinions welcome.
  robert


--


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[asterisk-users] Play a file on a channel from the Manager API

[GDrayer]

> Is there any way to play a file on a channel from the Manager API 
> (other than from Originate)?

This question was asked by someone else on the ast-dev list and the only
advice given was that Redirect was the solution.  I find myself with the
same problem now but I don't understand the response.

The situation: I need to play a file from the Asterisk Manager on a
channel that is currently in a call.  I don't want to break them out of
the call to play the message and I only want one specific channel to
hear the message.  In effect I want to ChanSpy the channel but to play a
message instead of speak to the person on the channel.

How does Redirect provide a solution?  

Thanks again,
George
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[asterisk-users] Asterisk and unicall + mfcr2 signalling

[Joca Loco]

Hi,

I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I
have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I
can make calls over E1, but can't receive.

Here the CLI when I make a call:

   -- Executing [EMAIL PROTECTED]:1] Dial("SIP/23-081cbc40",
"Unicall/g1/91642208|50") in new stack
   -- Called g1/91642208
[May 14 10:34:59] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Dialing
[May 14 10:34:59] NOTICE[4620]: chan_unicall.c:1959 unicall_exception:
Exception on 8, channel 1
[May 14 10:35:14] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Alerting
   -- Hungup 'UniCall/1-1'
 == Spawn extension (ps5, 006191642208, 1) exited non-zero on
'SIP/23-081cbc40'
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK
[May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Drop call
[May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/1 event Release call
   -- Unicall/1 released


And here the CLI when I receive a call:

[May 14 10:35:51] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/8 event Detected
[May 14 10:35:52] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event:
Unicall/8 event Protocol failure
[May 14 10:35:52] ERROR[2914]: chan_unicall.c:2603 handle_uc_event:
Unicall/8 protocol error. Cause 32772

Any idea why I can't receive calls, and fot Unicall protocol error Cause
32772?

Thanks, Joca Loco.
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[asterisk-users] Re: Mobile Number to Mobile carrier mapping

[Steven]

Not now that they have intoduced number portability.
The phone companies have to keep huge databases to keep track of which carrier 
to send the call to.

-- 
-- 
Steven

http://www.glimasoutheast.org



  "Ritesh Agrawal" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
  Hi Folks,

  Is there a way to find out the mobile/landline carrier name based on the 
phone number?
  For example, who is the mobile carrier for (415)2345678
  I had heard about some query but just don't remember how/what? 

  Thanks in advance.
  Ritesh





--


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[asterisk-users] Re: RE: Digital Phones

[Steven]

We use the Handset Gateways from Citel.

They convert SIP to Digital Handsets, so there is no hardware to add to the 
server and you can still use your 2-wire phone lines.

-- 
-- 
Steven

http://www.glimasoutheast.org



"bilal ghayyad" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Hi;

Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?

Regards
Bilal Ghayad




Bored stiff? Loosen up...
Download and play hundreds of games for free on Yahoo! Games.
http://games.yahoo.com/games/front
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Re: [asterisk-users] How to bring MoH volume down

[Alex Balashov]

Zeeshan,

On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect:

MoH volume is uncomfortably high and I want to bring it down. Its mpg123. 
How can I do it?


  There are some settings in musiconhold.conf that may yield the desired
effect:

[default]
mode=mp3
directory=/var/lib/asterisk/moh

; valid mode options:
; quietmp3  -- default
; mp3   -- loud
; mp3nb -- unbuffered
; quietmp3nb-- quiet unbuffered
; custom-- run a custom application
; files -- read files from a directory in any Asterisk supported 
format


  If not, it may come down to adjusting the base amplitude of the entire 
track down.  I don't think there's a way to modify the gain specifically

for MOH.

-- Alex

--
Alex Balashov   <[EMAIL PROTECTED]>
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Re: [asterisk-users] How to bring MoH volume down

[Matt]

Remix your wav/mp3 files with a lower volume :)

On 5/14/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:


Hi,

MoH volume is uncomfortably high and I want to bring it down. Its mpg123.
How can I do it?

--
Zeeshan A Zakaria
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[asterisk-users] Problem with queue

[gc]

Asterisk 1.2.17

I am starting  to have problem with one of my queue. Everytime when I try to 
login an agent with AgentCallBackLogin(), it will play periodic announcement 
for the queue during this function call. Also when this agent answer the call, 
during the conversation, the agent also hear the periodic announcement. I tried 
to delete the agent completely from the queue or recreate the queue, the 
problem still persist. I have not yet restart the asterisk because this is our 
production server.

Gary

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RE: [asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized

[Nabeel Jafferali]

Did you have the IP specified in sip.conf?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Yaakov Menken
> Sent: May 13, 2007 10:43 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized
> 
> Yesterday we moved one of our servers to a new IP. We updated DNS and
> various adapters configured to register to that server registered to
> the
> new IP correctly. All seemed to be well.
> 
> This evening I discovered that with one exception, all of the adapters
> are getting a SIP/2.0 401 Unauthorized message back from asterisk. The
> exception is an Innomedia adapter -- Linksys PAP2's and (I believe) one
> Cisco ATA-188 are getting the "Unauthorized".
> 
> I have stopped, restarted, unloaded & loaded sip, and erased astdb to
> start from scratch... no dice. None of the config files have changed,
> and, as I said, they all appeared to work last night.
> 
> Can anyone give me a clue here?
> 
> Yours,
> 
> Yaakov Menken
> 
> --
> Yaakov Menken
> Capalon Communications, Inc.
> Ask us about Voice over IP for Business!
> 
> http://www.capalon.com
> 888-CAPALON (227-2566)
> 410-358-9800 x120
> 410-510-1053 fax
> 443-413-1042 cell
> [EMAIL PROTECTED]
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Re: [asterisk-users] RTP Mixer

[Tzafrir Cohen]

On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote:
> Hi
> 
> Can somebody brief me the working of RTP mixer from MeetMe perspective.

(RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get 
mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer)

Aparantly people either don't know enough or don't have the time.

Try rephrasing your question so it will be more specific and thus also 
hopefully take shorter time to answer.

Do you have a working system? Do you need to set up one? What version 
of Asterisk? What types of channels do you try to mix?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Call to Skype network

[Hugo Miguel de Almeida Teixeira Picao]

Hi There,

Good guide on setting up chanskype on trixbox

http://www.geek-pages.com/articles/asterisk/setting_up_trixbox/asterisk_to_use_skype.html

also:

http://www.chanskype.com/

working on my trixbox 2.0 :)

Best Regards,

Com os melhores cumprimentos,

Hugo Picão

Link Consulting - Redes&Segurança

Tel: 213 100 182
Av. Duque de Ávila, 23
1000-138 Lisboa
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Bour
Sent: segunda-feira, 14 de Maio de 2007 12:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Call to Skype network

Open source...I wish...at least not to my knowledge yet.  Likely
something to do with the licensing for Skype...someone correct me here
if appropriate.

I'll drop a second email with details on the configuration unless
someone else pipes up requesting it.
D. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Verscheure
Sent: Sunday, May 13, 2007 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call to Skype network

yeah that would be great! Aren't there any open-source projects out
there who handle this?

greetz

2007/5/13, Dave Bour <[EMAIL PROTECTED]>:
> On x86 asterisk systems, there's 3 options out there, of which the 
> Chanskype one I've found to be the best.  It's $20 US for a single 
> channel personal license or $99 / per channel on a business license.  
> On the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to 
> configure it.  I've made a couple notes too if you want, I can send 
> offlist (unless it's generally wanted here onlist as I don't like 
> taking credit for others work).
> D
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tim 
> Verscheure
> Sent: Saturday, May 12, 2007 9:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call to Skype network
>
> Hi everyone,
>
> Is it possible to call from your Asterisk server to the Skype network?
> i.e., let's say I would like to call from an extension from my 
> Asterisk PBX machine to a Skype account, is this possible?
>
> I did a little bit of searching and they were talking about that's 
> only possible with windows machines, is this true?
>
>
> greetz, Tim
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RE: [asterisk-users] Dundi and unknown remote peers

[Asterisk]

Hmm, I tried this, but I get the following notice:

NOTICE[27486]: pbx_dundi.c:4695 set_config: Ignoring invalid EID entry
'*'


Do you perhaps know for any other option?

Thanks, Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: Friday, May 11, 2007 7:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Dundi and unknown remote peers

> Is it possible to allow remote peers to connect to your local DUNDi
> Asterisk box, even if you don't have them listed in the dundi.conf?

I seem to remember something in the sample config file about a [*] entry
being possible...

One would assume that would cover connections from undefined DUNDi
clients.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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