Re: [asterisk-users] Encrypted password for voicemail

[Tzafrir Cohen]

On Mon, Nov 27, 2006 at 05:12:19PM -0800, je . wrote:
> Thanks for the response Tzafrir. I meant
> voicemail.conf for the passwords of course - my
> mistake. Trying to ensure that if voicemail.conf is
> opened by an attacker that all the passwords are not
> readily available. By hashing them or encrypting them
> in a DB it's going to be much harder for an attacker
> to obtain access to the passwords.
> 
> The only way to encrypt the sending of passwords to
> the voicemail is by using SIP-TLS? 

Those are two conflicting goals. If you only save a hash of the
passowrd, as in /etc/shadow, you cannot reproduce the original password
from it in order to calculate "similar" hashes for chalange-and-response
authentication.

So do you want to protect from an eves-dropper or from a local attacker?
Anyway, at the current state of afairs, you get basically nothing. 

> (which is not yet
> in production stage?).

If we leave development issues aside and look at things you can use now:
use stunnel to provide SSL/TLS support for it?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Modprobe zaptel reports FATAL: Module zaptel not found

[kjcsb]

I am (unsuccessfully) trying to install zaptel (incl ztdummy - I don't have
any Digium hardware) on CentOS 4.

uname -r
2.6.9-42.ELsmp

Not sure how this relates to 2.6.9-42.0.3 (see below)

ln -s /usr/src/kernels/`uname -r` /usr/src/linux
ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6

cd /usr/src/zaptel-1.2.11*
make linux26
You do not appear to have the sources for the 2.6.9-42.ELsmp kernel
installed.

rpm -q kernel-devel
kernel-devel-2.6.9-42.EL
kernel-devel-2.6.9-42.0.3.EL

I don't understand why there are two kernel-devel packages installed

rpm -q kernel-smp-devel
kernel-smp-devel-2.6.9-42.0.3.EL

rm /usr/src/linux
rm /usr/src/linux-2.6
ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux
ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux-2.6

cd /usr/src/zaptel-1.2.11*
make linux26
make install
make config

/sbin/modinfo zaptel
modinfo: could not find module zaptel

find /lib/modules | grep zaptel
/lib/modules/2.6.9-42.0.3.ELsmp/extra/zaptel.ko

cd /usr/src/kernels
ls
2.6.9-42.0.3.EL-hugemem-i686  2.6.9-42.0.3.EL-i686  2.6.9-42.0.3.EL-smp-i686
2.6.9-42.EL-i686

Could anyone shed any light on what I've done wrong?

Thanks

Cameron

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Re: [asterisk-users] vm_change_password shell?

[Tzafrir Cohen]

On Mon, Nov 27, 2006 at 11:26:35PM -0800, je . wrote:
> In Asterisk 1.2.13 in app/app_voicemail.c, line 4700
> ext_pass_cmd is checked to decide whether to use
> vm_change_password or vm_change_password_shell to
> change a user's password for his voicemail account.
> 
> I wonder, what is the difference between
> vm_change_password and vm_change_password_shell - what
> is that shell? The only reference I found on the
> Internet was the following bug report:
> http://bugs.digium.com/view.php?id=7361 , but it's not
> very useful. Anywhere I can find a little more
> information on the difference between these two
> scenarios?

Use the Source, Luke.

http://svn.digium.com/svn/asterisk/branches/1.4/apps/app_voicemail.c

In it, you'll find:

static void vm_change_password_shell(struct ast_vm_user *vmu, char *newpassword)
{
char buf[255];
snprintf(buf,255,"%s %s %s 
%s",ext_pass_cmd,vmu->context,vmu->mailbox,newpassword);
if (!ast_safe_system(buf))
ast_copy_string(vmu->password, newpassword, 
sizeof(vmu->password));
}


ext_pass_cmd is not defined in that function and hence a global, and its 
value is taken from the configuration item "externpass".


The idea is that if the user has a better way of changing the password,
then the externpass script should be used to change the password.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel not found

[Tzafrir Cohen]

On Tue, Nov 28, 2006 at 09:22:18PM +1300, kjcsb wrote:
> I am (unsuccessfully) trying to install zaptel (incl ztdummy - I don't have
> any Digium hardware) on CentOS 4.
> 
> uname -r
> 2.6.9-42.ELsmp
> 
> Not sure how this relates to 2.6.9-42.0.3 (see below)
> 
> ln -s /usr/src/kernels/`uname -r` /usr/src/linux
> ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6

Unnecessary. Just install the relevant kernel-devel package. What
instructions are you following?

> 
> cd /usr/src/zaptel-1.2.11*
> make linux26
> You do not appear to have the sources for the 2.6.9-42.ELsmp kernel
> installed.

What is the output of:

uname -r

> 
> rpm -q kernel-devel
> kernel-devel-2.6.9-42.EL
> kernel-devel-2.6.9-42.0.3.EL
> 
> I don't understand why there are two kernel-devel packages installed
> 
> rpm -q kernel-smp-devel
> kernel-smp-devel-2.6.9-42.0.3.EL
> 
> rm /usr/src/linux
> rm /usr/src/linux-2.6
> ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux
> ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux-2.6
> 
> cd /usr/src/zaptel-1.2.11*
> make linux26
> make install
> make config
> 
> /sbin/modinfo zaptel
> modinfo: could not find module zaptel

If you run 'depmod', does it change anything?

> 
> find /lib/modules | grep zaptel
> /lib/modules/2.6.9-42.0.3.ELsmp/extra/zaptel.ko
> 
> cd /usr/src/kernels
> ls
> 2.6.9-42.0.3.EL-hugemem-i686  2.6.9-42.0.3.EL-i686  2.6.9-42.0.3.EL-smp-i686
> 2.6.9-42.EL-i686

-- 
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[asterisk-users] AGI Script with parameters

[Olivier Saulnier]

Hello,

I want to use AGI for give some information for a softphone, as:
exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
We use Ruby langage.
The line doesn't worksin as this, but works with shell command.
Also, if i modify my ruby script for give in the code the ip adress and 
port number, it works!


Do you have any idea??
best regards,

--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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[asterisk-users] Re: upgraded polycom to 2.0.1.0291 and...

[Shaun]

I corrected the problem.  I think it was a problem with the bootrom not 
being new enough, not sure because when you upgrade the bootrom it formats 
the phone so maybe it was configs.  I dont think so though because the 
configs i used where the same even after the bootrom but thats not to say 
that a config stored on the phone wasnt taking over...

Since then i've bumped up to 2.0.3 but 2.0.1 worked for me also once i did 
that.

-- 

~Shaun
"Noah Miller" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> Hi Shaun -
>
>> ERROR[4391]: chan_sip.c:11169 handle_request: Missing Cseq. Dropping this
>> SIP message, it's incomplete.
>>
>> I'm having to use the configs that came with the zip because apparently 
>> my
>> previous configs no longer are valid and lock the phone from dialing with 
>> a
>> url disabled message... anyway, these polycom phones are driving me 
>> crazy,
>> expecially their configs!
>
> The 2.x sip version has many options that are not available in the 1.x
> config files, so, no older config files won't work with newer
> firmwares.  Care to share your phone configs and any other pertinent
> information (asterisk version)?
>
> BTW: While some people are using 2.0.1 successfully (including me),
> the latest version from Polycom is 2.0.3.
>
> - Noah
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Re: [asterisk-users] AGI Script with parameters

[Anton Frolov]

Olivier Saulnier wrote:
> Hello,
> 
> I want to use AGI for give some information for a softphone, as:
> exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
> We use Ruby langage.
> The line doesn't worksin as this, but works with shell command.
> Also, if i modify my ruby script for give in the code the ip adress and
> port number, it works!
> 
> Do you have any idea??
> best regards,
> 

bonjour, Olivier

I think that your line should be something like:
exten => phone,2,agi,script.py|args

AF.
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Re: [asterisk-users] AGI Script with parameters

[Olivier Saulnier]

hello,

I try:
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10 5010

And it doesn't works again...
best regards,
Olivier S

Anton Frolov a écrit :


bonjour, Olivier

I think that your line should be something like:
exten => phone,2,agi,script.py|args
 




--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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Re: [asterisk-users] AGI Script with parameters

[Anton Frolov]

Olivier Saulnier wrote:
> hello,
> 
> I try:
> exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10 5010
> 
> And it doesn't works again...
> best regards,
> Olivier S

you should use a separator between the arguments as well. try
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010

AF.
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Re: [asterisk-users] Click to dial apps always show from "asterisk"

[Tim Panton]

On 28 Nov 2006, at 03:01, Eric Bishop wrote:

I am trying to do it with FOP and Calling Circles. Both have closed  
code. Anyway to do it from Asterisk?




You could use the 'Local' channel as the argument to the originate  
command

and then set it in the dialplan.

Tim Panton

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



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Re: [asterisk-users] AGI Script with parameters

[Tim Panton]

On 28 Nov 2006, at 08:47, Olivier Saulnier wrote:


Hello,

I want to use AGI for give some information for a softphone, as:
exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
We use Ruby langage.
The line doesn't worksin as this, but works with shell command.
Also, if i modify my ruby script for give in the code the ip adress  
and port number, it works!


Do you have any idea??
best regards,



We use a "?" :

exten => 1,n(agi),Agi(${AGI}?FORWARD_CALLCLASS_ID=$ 
{FORWARD_CALLCLASS_ID})



But we also do :

exten => 1,n,set(STRIKE2=NO)
; set more vars
exten => 1,n(agi),Agi(${AGI})

then do  a getVariable("STRIKE2") in the AGI

Tim.



http://www.westhawk.co.uk/industries/voip.html
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Re: [asterisk-users] AGI Script with parameters

[Olivier Saulnier]

YES!!
It works ;-))

Best regards,
Olivier S.

Anton Frolov a écrit :




you should use a separator between the arguments as well. try
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010

 



--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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[asterisk-users] Attn: DISA Experts(Strange problem with DISA)

[Crazy Boy]

Hi Friends,

I am facing a strange problem with DISA. I have installed and configured 
Trixbox. I've created a secret extension i.e., 555 and called this extension in 
Digital Receptionist using custom extension i.e., created in 
extensions_custom.conf file.

When I call from my mobile phone to my PSTN number, which is connected to FXO 
port, my IVR is responding. After entering my DISA extension(555), its asking 
password. After entering correct password, its giving ringtone. Upto this, no 
problem. The problem is coming here only. When I enter a USA number, its taking 
the first digit of USA number twice. 
For eg: If I enter 17187773456, its taking as: 117187773456
If I enter 917187773456, its taking as: 9917187773456 

Its taking my input USA number correctly for sometimes and call is connected to 
my mobile. I tried by changing the value of "relaxdtmf" from yes to no and vice 
versa.

Here I am sending my config files. Please tell me the solution.

Extenstions_custom.conf contents:

[custom-CLID]
exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,Authenticate(1234)
exten => s,5,DISA(no-password|disa-ext)

[disa-ext]
exten => _9.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN:1},90,tr)
exten => _9.,2,Hangup

Zapata.conf file contents:

[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300 

usecallerid=yes
relaxdtmf=yes
dtmfmode=rfc2833
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6

faxdetect=incoming
#include zapata-auto.conf

group=1

#include zapata_additional.conf

Please tell me the solution. Looking forward to your response. Thank you.

Regards,
Chandra.

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[asterisk-users] Use of VPNs

[Barry Fawthrop]

Hi all

Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
Since the QoS and general traffic of the Internet can never be 
predicted, would the implementation of a VPN between Client and VoIP 
Provider increase voice quality and/or security or is the converse true ?


Thanks
Barry
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Re: [asterisk-users] Use of VPNs

[Conrad Wood]

On Tue, 2006-11-28 at 07:18 -0500, Barry Fawthrop wrote:
> Hi all
> 
> Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
> Since the QoS and general traffic of the Internet can never be 
> predicted, would the implementation of a VPN between Client and VoIP 
> Provider increase voice quality and/or security or is the converse true ?

if you do it right then yes, you'll get extra security.
Voice quality is likely to degrade due to the added overhead. 
However, some people reported ISPs mucking about with VoIP traffic and
encapsulating that into say IPSec or OpenVPN traffic should help.
I run VoIP traffic over OpenBSD IPSec connections quite happily (in UK,
using Mistral/Netkonect DSL) and are confident they don't get
intercepted or abused.
I encapsulate on a seperate box, not on the asterisk box directly to
avoid cpu spikes and to be able to shape QoS in my VPN.

Conrad



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[asterisk-users] Different click2call?

[Darko]

Hello List,

We are deveoping apication/system based on PHP5, Postgre,Ajax,ect..
It should be compleate sistem for realystate agensy and road worers(agents)
and it will be distributed system.
We made very good inplementation based on asterisk and OSP for distributed
offices and it will be part of system (integrated).

We would like to implement some options in system:
Agent have  contact list with phonebook (clients and sellers of
proprieities) in aplication, agent have hard IP phone or dual-mode phone.
The Idea is when he/she click on number in contact in aplication, number
shuld be forwarder to asterisk and if is connected to other side agent phone
will ring.
If call faild asterisk should informe agent that ather side is busy.

Please can anyone give us BEST IDEA HOW TO develop this?

Sorry for my English

Best regards

Darko

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[asterisk-users] Symbian Softphone

[Marnus van Niekerk]

Anybody know of a SIP/IAX
softphone for Symbian Series 60?  (Apart from the builtin Nokia one!)

Tx

M

-- 

"Opportunity is missed by most people because it is
dressed in overalls and looks like work."

Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.




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Re: [asterisk-users] Manage Users in LDAP

[marvin horst]

On 11/27/06, Anil Ramsingh <[EMAIL PROTECTED]> wrote:


We use a windows ldap browser & editor called ldapeditor from
http://www.ldapeditor.com . Its the best free browser but it only runs on
m$ windows.

On 11/27/06, Steven Baker <[EMAIL PROTECTED] > wrote:

> Hello All,
> we are using asterisk+openldap. Do is there any easy way to manage users
> besides command line or the java "ldap browser?
>


we use http://phpldapadmin.sourceforge.net/
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Re: [asterisk-users] Symbian Softphone

[Klaus Darilion]

Marnus van Niekerk wrote:
Anybody know of a SIP/IAX softphone for Symbian Series 60?  (Apart from the 
builtin Nokia one!)


sillyant.com




Tx

M

--

"Opportunity is missed by most people because it is
dressed in overalls and looks like work."

Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.





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--
Klaus Darilion
nic.at

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RES: [asterisk-users] CTI

[Hernany Oliveira]

I need something like Call Manager.
I need to know how many agents is logged in, how many calls are on queues,
transfer calls, hang up calls, reports and so on.
Everything related to a Call Center operation.
 
I have been looking for and I did not find anything.


-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Matt Florell
Enviada em: segunda-feira, 27 de novembro de 2006 17:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] CTI

CTI is a pretty broad term, what exactly do you need to do?

Do you need to connect with another system?

What features do you need?

MATT---

On 11/27/06, Hernany Oliveira <[EMAIL PROTECTED]> wrote:
> Is there any cti for asterisk ??
> Where may I download it ??
>
> Thanks in advance
>
> Hernany
>
>
> --
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date:
27/11/2006
> 04:00
>
>
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-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date: 27/11/2006
04:00
 

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.430 / Virus Database: 268.14.19/555 - Release Date: 27/11/2006
18:09
 

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[asterisk-users] Zaptel drivers for Solaris?

[Frank Tarczynski]

I'm looking to build the zaptel drivers on a Solaris  10 X86 box.  I've 
found the driver source code on 
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted 
along with Asterisk 1.2.7.1  Does anyone know of a fresher version?  Is 
this code considered "somewhat ready for prime time use"?


Thanks,
Frank
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Re: [asterisk-users] Anybody used Asterfax?

[Klaus Darilion]

Hi!

I once tried Asterfax but it is really difficult to install and 
configure, and running openoffice in an xvfb session, together with the 
need for java is IMO overkill. I'm now using hylafax+iaxmodem with 
virtual printer devices for windows clients and hy-email2fax for clients 
which prefer to send faxes by email (PDF attachment).


regards
klaus



Zeeshan Zakaria wrote:

Trying to install asterfax-1.1-freeb2.i386.rpm, I get following error. How
can I get rid of it.

Installing jre
Installing libtiff
Installing ghostscript
Installing Xvfb
Installing openoffice.org
Installing spandsp
Installing spandsp0.0.3
Spandsp did not install correctly.
error: %post(asterfax-1.1-freeb2.i386) scriptlet failed, exit status 1




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--
Klaus Darilion
nic.at

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Re: [asterisk-users] Can anyone enlighten me as to what this means?

[Rob McKrill]

You could use Eric's solution to minimize glare, but in my experience having
the circuit set to CO Yields causes a ton of problems.  This is only from my
experience with a couple of my customers and also my old PRI at the office.
Just about the time I figured out the problem with my circuit was glare we
ended up moving to Level(3)'s hosted product and disconnected the circuit.
Once 3tone went away we ended up getting a PRI through a different provider
and I made it a point to double and triple check the glare setting with the
provisioner.

We experienced "All Circuits Busy" messages on people calling into the
office.  We never had more than 8-10 calls going at one time when we were
receiving this message so there were plenty of channels available, but since
the CO was yielding to the CPE (CO Yields) it would immediately give that
"All Circuits Busy" message when it collided with our equipment.  I just
wanted to make it clear that these problems were with a different phone
system, not Asterisk.

I really suggest you have the phone company switch you to CPE Yields and
implement what Eric suggests as that is good practice anyway.


On 11/27/06, Matt <[EMAIL PROTECTED]> wrote:


Can I set the yield option on asterisk?  The telco confirmed it is set
to CO Yields... which to me doesn't make a whole lot of sence...

On 11/22/06, Rob McKrill <[EMAIL PROTECTED]> wrote:
> Check with your telco on the "Glare" setting.  They probably have Glare
set
> to "CO Yields" which tells their switch to 'yield' to your switch/pbx
when
> negotiating which channel to use.  We ran into this problem with our PRI
and
> an older phone system that did not give nearly the amount of insight
(logs)
> as to what the problem might be.
>
> It sounds like you probably want them to set it to "CPE Yields".
>
>
> On 11/21/06, Matt <[EMAIL PROTECTED]> wrote:
> >
> > We are doing PRIs into T4XXP cards.   When I call out things are
> > fine... however tonight sometimes on inbound calls I'd get:
> >
> > chan_zap.c: Duplicate setup requested on channel 0/1 already in use on
> span 1
> >
> > in the full debug log followed by a fast busy signal on the calling
> parties end.
> >
> > Anyone know what would cause that?
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re: [asterisk-users] AsteriskNow console access

[James Willing]

"Geoff Karl" <[EMAIL PROTECTED]> wrote:

> I just downloaded and installed the AsteriskNow appliance
> (http://www.asterisknow.org) .  This looks like it has lots of
> promise.

> Anyone know what the secret is to being able to actually login to the
> root console?

Yes, as I found out (rather painfully) after the second (or was it third)
install, for console access you have to login as 'admin', using the
password you entered during the installation.

And I agree that it looks promising, though as far as I can tell so far
none of the GUI functionality actually works yet.

So far, I have been unable to actually get it to commit any changes
entered via the GUI and after a few attempts (or an hour or so of running)
the GUI generally appears to stop functioning.  No response to 'system
info' selection, etc...

...but it is 'Beta 1' afterall...   B^}

-- 
-jim (Willing)
Midwest Connections, Inc.

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Re: [asterisk-users] Anybody used Asterfax?

[Zeeshan Zakaria]

Is it free, if not what's the cost and where can I get it from?
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Re: [asterisk-users] Voicemail, SQL & ODBC

[Norbert Zawodsky]

RR wrote:

> Mate, I can't say it with authority but I'm almost certain that the
> only DB that a specific driver was written for is MySQL. I think if
> you use res_mysql.o you should be able to talk to mySql directly
> without needing ODBC.



O.k., Nice to hear. But I'm not sure *how* to "use res_mysql.o".

In other words:

I configured cdr_mysql.conf and asterisk happily writes CDRs into my
mySQL DB on a different server. But how should I do that regarding
voicemail?

Norbert

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[asterisk-users] aastra 480i xml interface for "comedian" mail

[marvin horst]

Does anyone have an xml interface to comedian mail setup for the aastra 480i
CT?

The old interface to comedian mail on the ADSI phones was a nice feature
although kind of slow loading. I thought I'd try to duplicate it on the SIP
phones, but didn't want to reinvent the wheel.

Marv Horst
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Re: [asterisk-users] Zaptel drivers for Solaris?

[Steve Kennedy]

On Tue, Nov 28, 2006 at 08:30:55AM -0500, Frank Tarczynski wrote:

> I'm looking to build the zaptel drivers on a Solaris  10 X86 box.  I've 
> found the driver source code on 
> https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted 
> along with Asterisk 1.2.7.1  Does anyone know of a fresher version?  Is 
> this code considered "somewhat ready for prime time use"?

I thought it was for Solaris/Sparc anyway.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] Zaptel drivers for Solaris?

[Matt Florell]

The code from there is stable, and it has a lot of bugs fixed on it
that are fixed in later versions of the Asterisk 1.2 tree. The reason
it was forked is that the Asterisk developer community doesn't test
each release on Solaris and they put a lot of things into the code
that just plain won't work on Solaris, but work fine on Linux.

I have tested the code(on x86, not sparc) and it is very stable, and
the maintainers of the http://solarisvoip.com/ site are great to work
with if you find bugs or any issues with it not working with your
system.

MATT---

On 11/28/06, Frank Tarczynski <[EMAIL PROTECTED]> wrote:

I'm looking to build the zaptel drivers on a Solaris  10 X86 box.  I've
found the driver source code on
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
along with Asterisk 1.2.7.1  Does anyone know of a fresher version?  Is
this code considered "somewhat ready for prime time use"?

Thanks,
Frank
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Re: [asterisk-users] CTI

[Matt Florell]

Have you looked at QueueMetrics?
http://queuemetrics.loway.it/

There are also several call center packages for Asterisk out there
that have all of the reports built into them that you want:
http://www.voip-info.org/wiki/view/Predictive+dialer

MATT---

On 11/28/06, Hernany Oliveira <[EMAIL PROTECTED]> wrote:

I need something like Call Manager.
I need to know how many agents is logged in, how many calls are on queues,
transfer calls, hang up calls, reports and so on.
Everything related to a Call Center operation.

I have been looking for and I did not find anything.


-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Matt Florell
Enviada em: segunda-feira, 27 de novembro de 2006 17:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] CTI

CTI is a pretty broad term, what exactly do you need to do?

Do you need to connect with another system?

What features do you need?

MATT---

On 11/27/06, Hernany Oliveira <[EMAIL PROTECTED]> wrote:
> Is there any cti for asterisk ??
> Where may I download it ??
>
> Thanks in advance
>
> Hernany
>
>
> --
> No virus found in this outgoing message.
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Re: [asterisk-users] Symbian Softphone

[Marnus van Niekerk]

Klaus Darilion wrote:

sillyant.com
Thanx, but AFAIK that can only be used with their service, not asterisk 
or other SIP server.


M
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[asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.

[Jean-Marc Salsa]

Hi,

I have tried to use the Record Command in Asterisk,

Here is the configuration :
exten => record,1,Answer
...
exten => record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
exten => record,n,Playback(vm-goodbye)
exten => record,n,system(/usr/local/bin/send-recording.pl --to ${EMAILADDR}
--file /var/spool/asterisk/record/${CALLFILENAME}.WAV)
exten => record,n,Hangup

If I hung up the phone during recording,
then Message is well there, but Asterisk does not continue its way to the
system command to send me the file.

If I change the record command to detect a 2 sec silence.
Then, Asterisk hangs up correctly and DO send the file ...

Has anyone noticed something similar ?
Did I do something wrong ?
Can somebody help me ?

Thanks,

Jean-Marc
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Re: [asterisk-users] How to kill a meet me room at midnight

[Richard Lyman]

Eric Bishop wrote:


You can't hanup channels with a call file you can only create them no?


*snipped

actually you could hangup a call using a call file

example

Channel:  Tech/Dev-occurance
Application: Hangup
Data: somecausecodevar or digit equiv

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RE: [asterisk-users] Do extra CPU's help?

[Colin Anderson]

In 2.6.15 kernels and higher, you can use "taskset" to pin a task to a
certain CPU. Here's a way to set httpd to the 2nd processor in a 4 way
system:
 
HTTPDPID=`ps -A | grep -a -A0 "httpd"`
taskset 0x0002 -p  ${HTTPDPID:0:5}


-Original Message-
From: Don [mailto:[EMAIL PROTECTED]
Sent: Monday, November 27, 2006 10:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do extra CPU's help?


yes...a lot...especially in high volume situations...or situations with a
lot of transcoding

- Original Message - 
From: Eric   Bishop 
To: Asterisk Users Mailing List -  
Non-Commercial Discussion 
Sent: Tuesday, November 28, 2006 12:42 AM
Subject: Re: [asterisk-users] Do extra CPU's help?

Do extra CPU's without hyperthreading help?


On 11/28/06, Don < [EMAIL PROTECTED]  >
wrote: 

hyperthreading screws ours up...we actually run better with hyperthreading
off...
hyperthreading results seem to vary from different people you talk too.


- Original Message - 
From: Eric Bishop   
To: Asterisk   Users Mailing List -
Non-Commercial Discussion 
Sent: Monday, November 27, 2006 10:54 PM
Subject: [asterisk-users] Do extra CPU's help?

Hi all,

We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360).
We are seeing high load on multiple meetme session as well as g729
transcoding. My question is will putting an extra CPU help or does Asterisk
just run on a single CPU. 






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RE: [asterisk-users] Manage Users in LDAP

[Ejay Hire]

I second the vote for ldapadmin.
 
You can extend it with custom templates for your asterisk specific
attributes.
 
-ejay

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of marvin horst
Sent: Tuesday, November 28, 2006 6:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Manage Users in LDAP


On 11/27/06, Anil Ramsingh <[EMAIL PROTECTED]> wrote: 

We use a windows ldap browser & editor called ldapeditor from
http://www.ldapeditor.com   . Its the best free
browser but it only runs on m$ windows. 


On 11/27/06, Steven Baker <  
[EMAIL PROTECTED] > wrote: 


Hello All, 
we are using asterisk+openldap. Do is there any easy way to manage users
besides command line or the java "ldap browser?


we use http://phpldapadmin.sourceforge.net/ 
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Re: [asterisk-users] Symbian Softphone

[Emil Thelin]

On Tue, 28 Nov 2006, Marnus van Niekerk wrote:


Klaus Darilion wrote:
Thanx, but AFAIK that can only be used with their service, not asterisk or 
other SIP server.


AFAIK there is no "generic" sip-client for Symbian that you can use to 
connect to any other sip-service than the service that provides the 
client.


TiVi, Truphone are the ones I know about. TiVi works if you don't mind 
some lag. Haven't tried truphone yet as Im still waiting for it to work on 
my n80..


/e

--
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Re: [asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved

[John Reynolds]

Max,

How did you fix it? It seems like knowledge that could be shared, even if it
was basic oversight.

John


On 11/25/06, Max Bergmann <[EMAIL PROTECTED]> wrote:


Max Bergmann schrieb:
>
>
> How can i programming a Cisco 7961 to be used as busy lamp field?
>
> my configs :
>
> sccp.conf :
>
> [devices]
> type= 7961
> tzoffset= 0
> autologin   = 601
> speeddial   = *31, Hanna  --> other SIP telefon
>
> extensions.conf :
>
> exten => *31,hint,SIP/hanna
> exten => *34,hint,SCCP/601
>
>
> on SIP Telefon ( SNOM 360 ) everything functions good and i have busy
> lamp when cisco telefon Offhook, but differently does not function
> any idea ?
>
> Any input is greatly appreciated.
>
>
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I have solved my problem, thank you for your help


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Re: [asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.

[Anselm Martin Hoffmeister]

Am Dienstag, den 28.11.2006, 17:23 +0200 schrieb Jean-Marc Salsa:
> Hi,
>  
> I have tried to use the Record Command in Asterisk,
>  
> Here is the configuration :
> exten => record,1,Answer
> ...
> exten =>
> record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
> exten => record,n,Playback(vm-goodbye)
> exten => record,n,system(/usr/local/bin/send-recording.pl --to
> ${EMAILADDR} --file /var/spool/asterisk/record/${CALLFILENAME}.WAV) 
> exten => record,n,Hangup
>  
> If I hung up the phone during recording,
> then Message is well there, but Asterisk does not continue its way to
> the system command to send me the file.
>  
> If I change the record command to detect a 2 sec silence.
> Then, Asterisk hangs up correctly and DO send the file ...
>  
> Has anyone noticed something similar ?
> Did I do something wrong ?
> Can somebody help me ?

For a first workaround, use the hash key "#" to stop recording; afterwards, 
asterisk
will continue in the dialplan.

The moment that you hangup your phone, your dialplan will be aborted.

You might as well see wether the "h" extension could help. Read the fine
manuals at
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
and
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
for more information ;)

Anselm

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Re: [asterisk-users] Manage Users in LDAP

[Walt Reed]

I also use ldapadmin, but for many common tasks, I use custom command
line scripts that wrap standard ldap commands. I also wrote a couple
simple CGI's that allow users to change their password, select their
preffered shell, update GECOS, and a few other options. If you are
managing Asterisk users in LDAP too, I would imagine that a custom CGI
for managing asterisk specific attributes would be very useful. Yes,
templates within ldapadmin can also work, but sometimes you want
something more appropriate for end-users.

On Tue, Nov 28, 2006 at 09:29:50AM -0600, Ejay Hire said:
> I second the vote for ldapadmin.
>  
> You can extend it with custom templates for your asterisk specific
> attributes.
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[asterisk-users] Asterisk "Generating" SIP 486

[Jonathan Palley]

Hello -
 Desperately hoping someone has seen this before or can help:

In my Queue, Asterisk seems to be coming up with SIP 486 responses:

Called SIP/trainer2
   -- Got SIP response 486 "Busy Here" back from 221.219.11.71
   -- SIP/trainer2-08fae890 is busy


These are *definitely* not coming from the client!  Ethereal does not pick
up any 486 packets and is not indicating it is sending them.  It gets the
200 OK and that's it.  See the debug below:
  -- Called SIP/trainer2
qualify*CLI>
<-- SIP read from 221.219.11.71:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK52d58d56;rport
From: "asterisk" ;tag=as7c87f250
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Contact: 
User-Agent: Asterisk PBX
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
qualify*CLI>
<-- SIP read from 221.219.11.71:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK73ef8a87;rport
From: "IT726" ;tag=as7c139198
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: 
User-Agent: Asterisk PBX
Content-Length: 0


--- (9 headers 0 lines)---
   -- Got SIP response 486 "Busy Here" back from 221.219.11.71
Transmitting (NAT) to 221.219.11.71:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK73ef8a87;rport
From: "IT726" ;tag=as7c139198
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
   -- SIP/trainer2-08aa7af0 is busy


Any ideas/help would be highly appreciated.

Thanks,
Jonathan


--
Jonathan Palley | Idapted Inc.
[EMAIL PROTECTED]
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Re: [asterisk-users] Zaptel drivers for Solaris?

[Jason Parker]

I would say that whatever the solarisvoip svn repository has, is the latest 
version of Zaptel for Solaris.  If you didn't require Zaptel, you would be able 
to use Asterisk 1.4 (stock).  It's a trade off for now, I guess.  The main 
problem here, is that for kernel modules, the interface to the kernel is 
(usually) completely different between OSes.

- Matt Florell <[EMAIL PROTECTED]> wrote:
> The code from there is stable, and it has a lot of bugs fixed on it
> that are fixed in later versions of the Asterisk 1.2 tree. The reason
> it was forked is that the Asterisk developer community doesn't test
> each release on Solaris and they put a lot of things into the code
> that just plain won't work on Solaris, but work fine on Linux.
> 
> I have tested the code(on x86, not sparc) and it is very stable, and
> the maintainers of the http://solarisvoip.com/ site are great to work
> with if you find bugs or any issues with it not working with your
> system.
> 
> MATT---
> 
> On 11/28/06, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
> > I'm looking to build the zaptel drivers on a Solaris  10 X86 box. 
> I've
> > found the driver source code on
> > https://svn.sunlabs.com/svn/solaris-asterisk but this source is
> posted
> > along with Asterisk 1.2.7.1  Does anyone know of a fresher version? 
> Is
> > this code considered "somewhat ready for prime time use"?
> >
> > Thanks,
> > Frank
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-- 
Jason Parker
Digium

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Re: [asterisk-users] AgentCallbackLogin deprecated?

[Jason Parker]

Yes, AgentCallbackLogin is deprecated, but it will not be removed until after 
1.4. 

- Original Message - 
From: Miguel Paolino <[EMAIL PROTECTED]> 
To: asterisk-users@lists.digium.com 
Sent: Monday, November 27, 2006 7:19:44 AM GMT-0600 US/Central 
Subject: [asterisk-users] AgentCallbackLogin deprecated? 

I would like to know if AgentCallbackLogin will be discontinued anytime shortly 
in Asterisk 1.4.x . I've read a page in voip-info that said so [1], but it was 
not an official announcement. I have to be sure, because I'm in the process of 
setting up a medium-to-big callcenter with Asterisk and calling the agents when 
a call is placed in a queue is a requisite (I've also read it can be done with 
the dialplan, but this app eases the work). 

[1] - http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin 
-- 
Regards, 

Miguel Paolino 

-- 
Jason Parker 
Digium 
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[asterisk-users] SIP ATA Device Problems

[Jerry Rasmussen]

I have an MG3 SIP ATA. This sip phone is registered and I am able to call the 
phone from another softphone. However, I am unable to place a call from the 
phone.

In addition, after calling the SIP ATA phone the sip phone does not see to hang 
up the call in complete. Can anyone shed some light on this problem.

Thanks
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Re: [asterisk-users] AgentCallbackLogin deprecated?

[Gavin Hamill]

On Tue, 28 Nov 2006 10:27:27 -0600 (CST)
Jason Parker <[EMAIL PROTECTED]> wrote:

> Yes, AgentCallbackLogin is deprecated, but it will not be removed
> until after 1.4. 

Is there an isolated example somewhere of how to use existing dialplan
logic and dynamic queue membership to simulate the current behaviour?

What about generation of statistics for callcentre monitoring? If this
is not taking place through chan_agent, won't it be reinventing the
wheel to have to simulate this behaviour, too?

Cheers,
Gavin.
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Re: [asterisk-users] Encrypted password for voicemail

[jezzzz .]

I was wondering if we could protect against both.
Sending a password encrypted would protect against
eavesdropping. Once the password has been received,
the hash of it is taken and compared with the hash of
the password saved, so it also takes care of a local
attacker.

I could certainly use SSL/TLS, but that still doesn't
take care of a local attack to obtain the passwords of
the users.

Thanks

Jez

--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> On Mon, Nov 27, 2006 at 05:12:19PM -0800, je .
> wrote:
> > Thanks for the response Tzafrir. I meant
> > voicemail.conf for the passwords of course - my
> > mistake. Trying to ensure that if voicemail.conf
> is
> > opened by an attacker that all the passwords are
> not
> > readily available. By hashing them or encrypting
> them
> > in a DB it's going to be much harder for an
> attacker
> > to obtain access to the passwords.
> > 
> > The only way to encrypt the sending of passwords
> to
> > the voicemail is by using SIP-TLS? 
> 
> Those are two conflicting goals. If you only save a
> hash of the
> passowrd, as in /etc/shadow, you cannot reproduce
> the original password
> from it in order to calculate "similar" hashes for
> chalange-and-response
> authentication.
> 
> So do you want to protect from an eves-dropper or
> from a local attacker?
> Anyway, at the current state of afairs, you get
> basically nothing. 
> 
> > (which is not yet
> > in production stage?).
> 
> If we leave development issues aside and look at
> things you can use now:
> use stunnel to provide SSL/TLS support for it?


 

Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
http://new.mail.yahoo.com
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Re: [asterisk-users] AGI Script with parameters

[Steve Edwards]

I write my AGI's in C and use the standard command line parser 
"getopt_long()" to parse the options passed to the AGI. It looks a little 
"wordy" but there is little doubt as to what is going on:


exten = s,n,agi(play-path,--debug,--path=${PROMO-PATH})

"getopt_long" does all of the heavy lifting and handles the options in any 
order and allows abbreviations. Depending on how you handle the AGI 
environment, you can even do some debugging from your shell command line.


5 years from now, which one is more "obvious?"

agi,/ruby/ring.rb|192.168.0.10|5010|10|y

or


agi(/ruby/ring.rb,--host=192.168.0.10,--port=5010,--timeout=10,--continue-on-fail=yes)

On Tue, 28 Nov 2006, Olivier Saulnier wrote:


YES!!
It works ;-))

Best regards,
Olivier S.

Anton Frolov a écrit :




you should use a separator between the arguments as well. try
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010




--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound

[kjcsb]

ln -s /usr/src/kernels/`uname -r` /usr/src/linux
ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6


Unnecessary. Just install the relevant kernel-devel package. What
instructions are you following?

I already had kernel-devel installed and was still getting the message "You 
do not appear to have the sources for the 2.6.9-42.ELsmp kernel installed". 
So I hunted around and found a link indicating that *possibly* I needed to 
create a symbolic link. I tried this in two ways:

ln -s /usr/src/kernels/`uname -r` /usr/src/linux etc
Got the same message at make linux26

ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux etc
make linux26 was successful but modprobe can't find zaptel




What is the output of:
uname -r

2.6.9-42.ELsmp




If you run 'depmod', does it change anything?


No

Cameron

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[asterisk-users] AGI and some informations

[Olivier Saulnier]

Hello,

i would like to use AGI and Ruby for communicate with a softphone. I 
would like send the IP adress of the softphone, directly for the 
extensions.conf file, as:

exten => 302,1,agi,/ruby/ipphone.rb|ip_adress

I know, with extensions.conf file, that i work on the softphone 302. 
But, how can i read the ip adress (the softphone give me when it  
registered, but after???


best regards,

--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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Re: [asterisk-users] Different click2call?

[Tim Panton]

On 29 Oct 2006, at 11:30, Darko wrote:


Hello List,

We are deveoping apication/system based on PHP5, Postgre,Ajax,ect..
It should be compleate sistem for realystate agensy and road worers 
(agents)

and it will be distributed system.
We made very good inplementation based on asterisk and OSP for  
distributed

offices and it will be part of system (integrated).

We would like to implement some options in system:
Agent have  contact list with phonebook (clients and sellers of
proprieities) in aplication, agent have hard IP phone or dual-mode  
phone.
The Idea is when he/she click on number in contact in aplication,  
number
shuld be forwarder to asterisk and if is connected to other side  
agent phone

will ring.
If call faild asterisk should informe agent that ather side is busy.

Please can anyone give us BEST IDEA HOW TO develop this?


I'm biased, but take a look at Corraleta - it does both
classic "click to call" as you described and also
"click to call 2.0" where the call originates in a browser based
softphone.

See the url below.


Tim Panton

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



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Re: [asterisk-users] Zaptel drivers for Solaris?

[Andrew Joakimsen]

Solaris has poor support for anything in general. The Solaris version of
asterisk is outdated, and doesnt even compile correctly. Is one individual's
flawed testing your basis to use Asterisk on Solaris?

On 11/28/06, Frank Tarczynski <[EMAIL PROTECTED]> wrote:


I'm looking to build the zaptel drivers on a Solaris  10 X86 box.  I've
found the driver source code on
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
along with Asterisk 1.2.7.1  Does anyone know of a fresher version?  Is
this code considered "somewhat ready for prime time use"?

Thanks,
Frank
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Re: [asterisk-users] AGI and some informations

[Anton Frolov]

why don't you want to manage everything in Ruby?
I would call the script with just the number called
exten => 302,s,agi,script,${EXTEN}
and then make all of the decisions based on the phone number inside of Ruby.
When registering a softphone, you could store its IP in a database. Then
your script will take this IP from the database and use it.

When registering the softphone:
SoftPhonesDB.insert("olivier", $ip);

In extensions.conf:
exten => 302,s,agi,script.rb,${EXTEN}

In script.rb:
$ip = SoftPhonesDB.select("olivier");
Dial(SIP/$ip, $arg);

AF.


Olivier Saulnier wrote:
> Hello,
> 
> i would like to use AGI and Ruby for communicate with a softphone. I
> would like send the IP adress of the softphone, directly for the
> extensions.conf file, as:
> exten => 302,1,agi,/ruby/ipphone.rb|ip_adress
> 
> I know, with extensions.conf file, that i work on the softphone 302.
> But, how can i read the ip adress (the softphone give me when it 
> registered, but after???
> 
> best regards,
> 
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[asterisk-users] Return codes

[Doug Crompton]

How does one process a return code in Asterisk?

Example...

exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")


If the caller hangs up on the playback command the file remove System
statement after it never gets executed. The playback command returns a -1
in this case and logs a warning. The only thing mentioned in the command
reference is 101 for file not found.

Doug

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Re: [asterisk-users] Return codes

[Tim Panton]

On 28 Nov 2006, at 17:54, Doug Crompton wrote:


How does one process a return code in Asterisk?

Example...

exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")


If the caller hangs up on the playback command the file remove System
statement after it never gets executed. The playback command  
returns a -1
in this case and logs a warning. The only thing mentioned in the  
command

reference is 101 for file not found.


This isn't the answer to your question, but in that particular case,  
I'd put the rm into the 'h'

extension, that way it gets run whenever the user hangs up.

exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => h,1,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")


Tim Panton

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



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Re: [asterisk-users] Zaptel drivers for Solaris?

[Matt Florell]

For the record, it does compile correctly, even on x86 Linux(if you
change the architecture flag) and it runs very reliably as well on the
machines I have run it on.

For the Sparc platform this is the only real option you have to run
Asterisk, so for that it's not bad at all, for anyone on x86 Linux I
would recommend 1.2.13.

MATT---

On 11/28/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:

Solaris has poor support for anything in general. The Solaris version of
asterisk is outdated, and doesnt even compile correctly. Is one individual's
flawed testing your basis to use Asterisk on Solaris?

 On 11/28/06, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
> I'm looking to build the zaptel drivers on a Solaris  10 X86 box.  I've
> found the driver source code on
> https://svn.sunlabs.com/svn/solaris-asterisk but this
source is posted
> along with Asterisk 1.2.7.1  Does anyone know of a fresher version?  Is
> this code considered "somewhat ready for prime time use"?
>
> Thanks,
> Frank
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[asterisk-users] hang up detection

[Matic]

Hi,

I have a small problem with tdm2400 with tone detection. About 5% of all 
calls doesn't end because Asterisk dosen't detect "busy" tone on remote 
hangup. I know that board does support current reversal and current off, 
but how do I configure asterisk to use tone detection first and if that 
should fail, current off detection which my operator support (40 sec 
after remote hang up).


Matic

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[asterisk-users] WANTED : Zaptel Patch - Dtmfthreshold

[Stefan Agethen]

I am using Zaptel for my Wildcard and got DTMF Tones in my conversations 
since a long time.


To control this , i search for a possibility to control the detection of 
DTMF and to control talkoff, called "dtmfthreshold".


As seen in mISDN*  i am in need of an Patch to control the 
"dtmfthreshold" in zaptel, it must be possible, because there is an value,
staticly set to 1000, the value could be controlled between x and 4000, 
thats the things i "think to know".


There seems to be no other way than writing a patch to set this value in 
a config file.


So - is there anyone who could write such a method - my programming 
background is not the best to do this.


I would be very happy and - of course - there are many others who could 
use this option in the future.


I have tried MANY Solutions, so please - lets dont talk about relaxdtmf 
and so on ;)


--misdn.org

*mISDN -- copy of the FAQs found in misdn.org :

*Why are my dtmf tones not detected everytime?*

We've added a configurable *dtmftreshold*, the default is 100, it can 
have values between 20 and 500. It needs to be changed either by giving 
the module parametern dtmftreshold_option to the mISDN_dsp modul, or 
simply by setting the value


*dtmftreshold*=xxx

in the /etc/misdn-init.conf

--misdn.org

Greets from Germany to the Asterisk Community,

Stefan
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[asterisk-users] Best email client for Asterfax

[Olivier]

Hi,

Asterfex seems promising providing email to fax services.
With it, to send a fax to Mr Foo whose fax number is 123456789, an end user
just has to use its favorite email client and fill destination field (To:)
with [EMAIL PROTECTED]

I'm looking for an email client I could personnalize (configure ? use or
develop a plugin ?) so that end users would only have to type callee's name
and let the email client convert this name into the right fax number.

Is it possible to get close to this behaviour with Outlook or Outlook
Express ?
Ideally, you would have a new "Fax To:" field beside "To:".

Any hint ?
Regards
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Re: [asterisk-users] Encrypted password for voicemail

[Tzafrir Cohen]

On Tue, Nov 28, 2006 at 08:52:22AM -0800, je . wrote:
> I was wondering if we could protect against both.
> Sending a password encrypted would protect against
> eavesdropping. Once the password has been received,
> the hash of it is taken and compared with the hash of
> the password saved, so it also takes care of a local
> attacker.

Send an encypted password? Encrypted how, exactly? One common mistake is
to suggest to simply send the hash, as it is encrypted. But this merely
makes the hash a "password equivalent": An evesdroper can use the hash
to authenticate without knowing the password.

> 
> I could certainly use SSL/TLS, but that still doesn't
> take care of a local attack to obtain the passwords of
> the users.

-- 
   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] Modprobe zaptel reports FATAL: Module zaptel notfound

[Tzafrir Cohen]

On Wed, Nov 29, 2006 at 06:17:27AM +1300, kjcsb wrote:
> >>
> >>ln -s /usr/src/kernels/`uname -r` /usr/src/linux
> >>ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6
> >
> >Unnecessary. Just install the relevant kernel-devel package. What
> >instructions are you following?
> >
> I already had kernel-devel installed 

The right package?

What is the output of:

  ls -l /lib/modules/`uname -r`/build

-- 
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Re: [asterisk-users] hang up detection

[Tzafrir Cohen]

On Tue, Nov 28, 2006 at 07:13:38PM +0100, Matic wrote:
> Hi,
> 
> I have a small problem with tdm2400 with tone detection. About 5% of all 
> calls doesn't end because Asterisk dosen't detect "busy" tone on remote 
> hangup. I know that board does support current reversal and current off, 
> but how do I configure asterisk to use tone detection first and if that 
> should fail, current off detection which my operator support (40 sec 
> after remote hang up).

Just for the record: which telco is it? Doesn't it provide decent hangup
notification?

-- 
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icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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[asterisk-users] Why is * continually "destroying call"

[Larry Alkoff]

Looking at the CLI in Asterisk 1.2 it constantly reports "destroying 
call" and gives an address of any of 6 sip phones connected to it.


However, there are no calls made by anyone in the last hour
so why is it destroying calls?

The phones are all on fixed IP - fixed by my firewall by the MAC 
address.  You can see the last octet of the IP change.


Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Do extra CPU's help?

[Tzafrir Cohen]

On Tue, Nov 28, 2006 at 02:54:45PM +1100, Eric Bishop wrote:
> Hi all,
> 
> We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360).
> We are seeing high load on multiple meetme session as well as g729
> transcoding. My question is will putting an extra CPU help or does Asterisk
> just run on a single CPU.

Your question is not clear.

An extra processor will always help. Asterisk is a multithreaded
program. Things like transcoding are done in the threads of the specific
channels and will thus benefit nicely from extra CPUs.

I do suspect that Metme will not scale as well. Much of the Zaptel
conferencing code is done in the clock interrupt of the master zaptel
device, and that work would not be easily spread between several CPUs.
But I have not reviewed this well.


If you ask about single CPU vs. the 2 CPUs emulated by hyperthreading:
not exactly sure. I can speculate everal arguments in both directions.

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

[Vicky]

I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor contains
information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can
be a big mess if there are more than 1000-2000 files in that folder and very
hard to locate a call recording based on call time and extension number who
dialled. I need to put something like outgoing number dialled within call
file name instead of uniqueid .. After watching in console i  opened up
/var/lib/asterisk/agi-bin/recordingcheck and saw that it is setting
callfilename variable with extension number,time,unique id , etc. so i
edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables ) but its
just not giving dialed number and hence callfilename  doesnt contain
outgoing number . Any suggestions how can i get outgoing call number in
recording file ?
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Nicolas S.]

I didnt forgot the french translation, it's coming was just busy. 
In 1-2 weeks i ll provide it to you 

Regards


Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :
> Hi all!
> 
> We've released VoiceOne 0.4.0, a web-based and open source solution 
> which allows to fully manage an Asterisk service hosted on a LAMP server.
> 
> We focused on an charming and overall user-friendly interface. Thanks to 
> the authentication based on roles, once configured by a super user, the 
> PBX may be easily maintained even by an Asterisk unskilled users.
> 
>  From a technical point of view, the application is made up of two 
> modules: one for the client - i.e. the user interface - and the other 
> for the server. Thanks to the web services provided by the server module 
> and the use of a database, VoiceOne may be easily integrated with other 
> applications (e.g. CRM software).
> 
> The project has grown and has received positive response so far. 
> Nowadays there's a little but enthusiastic community of developers, 
> supporters and users. Translations in several languages (e.g. English, 
> Spanish, Russian, etc.) are already available.
> 
> On the project website at http://www.voiceone.it you'll find the online 
> demo and the links to download the source files from Sourceforge, as 
> well as a support forum.
> 
> We would be pleased if you could give it a try and let us know your 
> feedback, comments, ideas, or suggestions replying here or posting a 
> message on our forum.
> 
> Thanks for your kind attention.
> 
> Regards,
> Alex
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RE: [asterisk-users] Why is * continually "destroying call"

[Colin Anderson]

IIRC, a "call" from the SIP perspective is any transaction or interaction
with a SIP device. So things that qualify as a "call" are things like
registration and qualification. Nothing to sweat about. You can suppress it
with "sip no debug" from the command prompt. 

hth

-Original Message-
From: Larry Alkoff [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 28, 2006 12:04 PM
To: Asterisk-users
Subject: [asterisk-users] Why is * continually "destroying call"


Looking at the CLI in Asterisk 1.2 it constantly reports "destroying 
call" and gives an address of any of 6 sip phones connected to it.

However, there are no calls made by anyone in the last hour
so why is it destroying calls?

The phones are all on fixed IP - fixed by my firewall by the MAC 
address.  You can see the last octet of the IP change.

Larry
-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] hang up detection

[Matic]

Of course it does provide quite descent hangup notification (425Hz 200ms 
200ms envelope for 40s), but it is asterisk/digium that in about 5% 
fails to detect it. I have trie to alter Tx, Rx gain and I even change 
the busy counter to value 2. Normaly it does detect hangup in about 2 
seconds but for whatever reason in about 5% it fails to do so. After 
40sec of hangup notification telecom also provides current off 
notification but I donb't know how to instruct asterisk to also detect this.

.




Tzafrir Cohen pravi:

On Tue, Nov 28, 2006 at 07:13:38PM +0100, Matic wrote:
  

Hi,

I have a small problem with tdm2400 with tone detection. About 5% of all 
calls doesn't end because Asterisk dosen't detect "busy" tone on remote 
hangup. I know that board does support current reversal and current off, 
but how do I configure asterisk to use tone detection first and if that 
should fail, current off detection which my operator support (40 sec 
after remote hang up).



Just for the record: which telco is it? Doesn't it provide decent hangup
notification?

  


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Re: [asterisk-users] AGI and some informations

[Olivier Saulnier]

Anton Frolov a écrit :


When registering the softphone:
 


OK, in which file do i do that??


SoftPhonesDB.insert("olivier", $ip);

 

Could you explain me what means "SoftPhonesDB.insert"?? It's not an AGI 
commande, how can i use it??



In extensions.conf:
exten => 302,s,agi,script.rb,${EXTEN}

 


OK, i understand


In script.rb:
$ip = SoftPhonesDB.select("olivier");
Dial(SIP/$ip, $arg);

 

Hummm...Does Dial working with Ip adress?? Or should i write ine the 
database the softphone number too??


Best regards,
Olivier S

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Re: [asterisk-users] IAX access to FWD broken?

[Timothy Parez]

I've got the same problem here.
It can't register anymore --> timeout


Brian Capouch schreef:
I hadn't used FWD for quite a while.  A customer sent me an email last 
week, "Is FWD broken when one tries to use it with IAX?"


I have been playing around, and indeed seems to be the case.

Is there anyone out there successfully using the two of them together?

Thanks.

B.



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Re: [asterisk-users] AGI and some informations

[Anton Frolov]

it was not a real code, but just a schema.
I can't write a more precise snippet of code, since I'm completely
unaware of your configuration.
But in any case, I would delegate all of the logic to your Ruby script
and keep the minimum in the extensions.conf.

AF.


Olivier Saulnier wrote:
> Anton Frolov a écrit :
> 
>> When registering the softphone:
>>  
>>
> OK, in which file do i do that??
> 
>> SoftPhonesDB.insert("olivier", $ip);
>>
>>  
>>
> Could you explain me what means "SoftPhonesDB.insert"?? It's not an AGI
> commande, how can i use it??
> 
>> In extensions.conf:
>> exten => 302,s,agi,script.rb,${EXTEN}
>>
>>  
>>
> OK, i understand
> 
>> In script.rb:
>> $ip = SoftPhonesDB.select("olivier");
>> Dial(SIP/$ip, $arg);
>>
>>  
>>
> Hummm...Does Dial working with Ip adress?? Or should i write ine the
> database the softphone number too??
> 
> Best regards,
> Olivier S
> 
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[asterisk-users] channel ending with /n

[Nuria Fernández Mingo]

Hi for all, 
I'm doing test with Asterisk and I have a question.
I've seen that exist differences between executing an AMI command (originate) 
with channel ending in /n, but what's the meaning?

-- 
Nuria
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[asterisk-users] Bad Voice Quality - IAX2 redirect

[hugolivude]

Asterisk 1.2.7
RedHat 9.0

Hi,
I've run into some voice degradation problems with IAX2:

I frequently have calls come in on a DiD provided by an ITSP.  I often
have to redirect these calls back out to the PSTN (i.e. to a cell
phone).  When this happens,  I don't want my server in the media path,
I want to hand it off to my ITSP instead and let them handle both ends
of the call.  I've been very careful to avoid using t or T in my dial
commands.

I couldn't get this to work with SIP (I'm behind a NAT BTW).  No
matter how I tried, the media continued to pass through my server, so
I switched from SIP to IAX2.

I've had much better success redirecting calls back to the PSTN using
IAX2.  I can see the handshakes in the CLI and once the redirected
call had been established, I can phyically disconnect my * server from
the Ethernet and the call is unaffected.

Unfortunately I'm getting complaints about call quality (I use ulaw
the whole way).

I don't think the problem can possibly be on my server or its
configuration given that the call is completly handed off as described
above.  Surely this must be the ITSP's problem, but perhaps I'm
missing something?  Any suggestions are welcome!

Thanks,
H
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Re: [asterisk-users] Bad Voice Quality - IAX2 redirect

[Andrew Joakimsen]

I have seen countless problems resolved by using "notransfer=yes" in
IAX.conf stuff like dropped calls, poor quality and even 1 way audio.

On 11/28/06, hugolivude <[EMAIL PROTECTED]> wrote:


Asterisk 1.2.7
RedHat 9.0

Hi,
I've run into some voice degradation problems with IAX2:

I frequently have calls come in on a DiD provided by an ITSP.  I often
have to redirect these calls back out to the PSTN (i.e. to a cell
phone).  When this happens,  I don't want my server in the media path,
I want to hand it off to my ITSP instead and let them handle both ends
of the call.  I've been very careful to avoid using t or T in my dial
commands.

I couldn't get this to work with SIP (I'm behind a NAT BTW).  No
matter how I tried, the media continued to pass through my server, so
I switched from SIP to IAX2.

I've had much better success redirecting calls back to the PSTN using
IAX2.  I can see the handshakes in the CLI and once the redirected
call had been established, I can phyically disconnect my * server from
the Ethernet and the call is unaffected.

Unfortunately I'm getting complaints about call quality (I use ulaw
the whole way).

I don't think the problem can possibly be on my server or its
configuration given that the call is completly handed off as described
above.  Surely this must be the ITSP's problem, but perhaps I'm
missing something?  Any suggestions are welcome!

Thanks,
H
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Re: [asterisk-users] SIP ATA Device Problems

[Andrew Joakimsen]

Are you sure you have configured the "dial plan" or "digit map" correctly in
the device? If you do a "sip debug" is the phone talking and sending INVITE?

On 11/28/06, Jerry Rasmussen <[EMAIL PROTECTED]> wrote:


 I have an MG3 SIP ATA. This sip phone is registered and I am able to call
the phone from another softphone. However, I am unable to place a call from
the phone.

In addition, after calling the SIP ATA phone the sip phone does not see to
hang up the call in complete. Can anyone shed some light on this problem.

Thanks

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Re: [asterisk-users] AgentCallbackLogin deprecated?

[Octavio Ruiz (Ta^3)]

> > Yes, AgentCallbackLogin is deprecated, but it will not be removed
> > until after 1.4. 
> 
> Is there an isolated example somewhere of how to use existing dialplan
> logic and dynamic queue membership to simulate the current behaviour?

http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt

> What about generation of statistics for callcentre monitoring? If this
> is not taking place through chan_agent, won't it be reinventing the
> wheel to have to simulate this behaviour, too?

Why? Seems that reinventing the well was the agentcallbacklogin
implementation, when it could be happend in dialplan logic.


-- 
  May I ask a question?
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[asterisk-users] No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone

[Vincent Delporte]

Hi

I have the following setup to make outgoing calls:

X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work 
behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone.


I just tried calling my own cellphone, but there is no sound either way.

Here's what I did on the X-Lite at home in the Topology section:
IP address : Discover global address
STUN server : Discover server
Port used on local computer : Manually specify range 8000-8019

Here are the ports that I forwarded from my NAT router at home:
UDP 5060
UDP 3478 (STUN; needed?)
UDP 8000 to 8019

Is there something else I should do, either on my home setup or at work on 
the NAT router or Asterisk?


Thank you!

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Re: [asterisk-users] Asterisknow

[Dumpolid Exeplish]

hi,
i have tried to use asterisk now but it seems to me that it doesnt retain
configurations. have you experinced the same thing ??


On 11/26/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:


Hello,

Anyone saw asterisknow, ?

Regards

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Re: [asterisk-users] AsteriskNow console access

[Dumpolid Exeplish]

i had the same problem. the GUI stopped responding to configuration changes.

On 11/28/06, James Willing <[EMAIL PROTECTED]> wrote:



"Geoff Karl" <[EMAIL PROTECTED]> wrote:

> I just downloaded and installed the AsteriskNow appliance
> (http://www.asterisknow.org) .  This looks like it has lots of
> promise.

> Anyone know what the secret is to being able to actually login to the
> root console?

Yes, as I found out (rather painfully) after the second (or was it third)
install, for console access you have to login as 'admin', using the
password you entered during the installation.

And I agree that it looks promising, though as far as I can tell so far
none of the GUI functionality actually works yet.

So far, I have been unable to actually get it to commit any changes
entered via the GUI and after a few attempts (or an hour or so of running)
the GUI generally appears to stop functioning.  No response to 'system
info' selection, etc...

...but it is 'Beta 1' afterall...   B^}

--
-jim (Willing)
Midwest Connections, Inc.

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[asterisk-users] Re: Re: Re: Re: Rewriting caller ID from database?

[Vincent Delporte]

At 10:34 28/11/2006 -0700, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I have done simple ael2 script, tak doing lookup in asterisk database 
like: find full numer, if cidname isn't found, substract one digit from 
right  and try again, and so on


Thanks. If LookupCIDname doesn't come with its own feature, I'll add it to 
the script.


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[asterisk-users] Billing software with reseller accounts

[Guillermo Salas M.]

Hello,


Can you recommend a good billing software for asterisk that supports
reseller accounts? Will be better if it haves opensource licence.

Best regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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Re: [asterisk-users] Voicemail, SQL & ODBC

[Derek Whitten]

Norbert Zawodsky wrote:
> RR wrote:
> 
>> Mate, I can't say it with authority but I'm almost certain that the
>> only DB that a specific driver was written for is MySQL. I think if
>> you use res_mysql.o you should be able to talk to mySql directly
>> without needing ODBC.
> 
> 
> 
> O.k., Nice to hear. But I'm not sure *how* to "use res_mysql.o".
> 
> In other words:
> 
> I configured cdr_mysql.conf and asterisk happily writes CDRs into my
> mySQL DB on a different server. But how should I do that regarding
> voicemail?
> 
> Norbert
> 
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http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Voicemail


http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage












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Re: [asterisk-users] Voicemail, SQL & ODBC

[Noah Miller]

Hi Peder -


Is the storage of actual voicemail messages in a database still limited
to ODBC?  If so, why?


Yes.  Why? Nobody has developed a voicemail solution that directly
connects to a *SQL database for message storage.



And is the use of mySQL and ODBC at the same time still a bad idea?  If
so, why?


ODBC is just a connection to interface with a database.  You have to
run a database like mysql in order to use an ODBC driver.  Can you use
the same mysql database for asterisk configs and odbc voicemail
storage: yes.



I want to store all of my voicemail stuff in a database so that I can
give users web access to it, but I don't want to run web services on my
* server itself.  If it is all in a DB, I can have a web box and a
separate SQL box and none of it should affect *.


Yes, you can do this.  There are other ways to do this without using
database storage for voicemail (e.g. you could use normal file-based
storage for voicemail and have the webserver connect to the vm storage
on the * server via nfs).


- Noah
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Re: [asterisk-users] Zaptel drivers for Solaris?

[Derek Whitten]

Andrew Joakimsen wrote:
> Solaris has poor support for anything in general. 


maybe you are looking in the wrong places for support...

SOLARIS IS NOT LINUX



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Re: [asterisk-users] IAX access to FWD broken?

[jason]

last I had heard, pretty much all FWD accounts that were created in the 
past  year or so no longer work with IAX. Still don't know why.


Timothy Parez wrote:

I've got the same problem here.
It can't register anymore --> timeout


Brian Capouch schreef:
I hadn't used FWD for quite a while.  A customer sent me an email 
last week, "Is FWD broken when one tries to use it with IAX?"


I have been playing around, and indeed seems to be the case.

Is there anyone out there successfully using the two of them together?

Thanks.

B.



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--
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet


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Re: [asterisk-users] Zaptel drivers for Solaris?

[Andrew Joakimsen]

I'm not looking for support for anything. I've alredy tried twice, going as
far as buying a Sun server (now running Linux) to get Solaris to work with
Asterisk and all I was able to use was the outdated packages from Mr.
Bendin, would not compile even using gmake + all the other dependancies.


Regards,

Andrew

On 11/28/06, Derek Whitten <[EMAIL PROTECTED]> wrote:


Andrew Joakimsen wrote:
> Solaris has poor support for anything in general.


maybe you are looking in the wrong places for support...

SOLARIS IS NOT LINUX



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[asterisk-users] Re: newbie question-asterisk username/password

[blackwater dev]

Ok, I am looking through the iax.conf file now and see 'guest' with no
password and tried to add another but none of these seem to let me log from
my softphone.  I did restart asterisk each time.

Thanks!

On 11/28/06, blackwater dev <[EMAIL PROTECTED]> wrote:


I have asterisk installed and now want to try it out.  I installed the
sample files and downloaded  http://www.loudhush.ro/ for my mac to use as
the phone.  When registering, it wants my asterisk username/password.  How
do I set this up?

Thanks!

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[asterisk-users] accountcode= placement in zapata.conf

[Damon Estep]

Can anyone tell me if accountcode= should appear before or after the
channels definition that you want it to apply to?

 

I have several groups of channels (PRI) defined in Zapata.conf, and wish
to specify accountcode by group.

 

If I put the accountcode= at the end of Zapata.conf it applies to all
channels configured, if I put it at the end of each group section the
behavior is inconsistent.

 

How is accountcode parsed in Zapata.conf, does it apply to everything
before or after the setting appears? Can it change mid-file? How?

 

Here is the config I am working with;

 

group=2

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_cpe

channel=25-47

accountcode=

;channels 25-47 should have accountcode=

 

group=3

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_cpe

channel=49-71

accountcode=

;channels 49-71 should have accountcode 

 

group=4

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_net

channel=73-95

accountcode=1714

;channels 73-95 should have accountcode 1714

 

Thanks!

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Re: [asterisk-users] accountcode= placement in zapata.conf

[Andrew Joakimsen]

Everthing should be ABOVE the channel => line for that group of channels


On 11/28/06, Damon Estep <[EMAIL PROTECTED]> wrote:


 Can anyone tell me if accountcode= should appear before or after the
channels definition that you want it to apply to?



I have several groups of channels (PRI) defined in Zapata.conf, and wish
to specify accountcode by group.



If I put the accountcode= at the end of Zapata.conf it applies to all
channels configured, if I put it at the end of each group section the
behavior is inconsistent.



How is accountcode parsed in Zapata.conf, does it apply to everything
before or after the setting appears? Can it change mid-file? How?



Here is the config I am working with;



group=2

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_cpe

channel=25-47

accountcode=

;channels 25-47 should have accountcode=



group=3

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_cpe

channel=49-71

accountcode=

;channels 49-71 should have accountcode 



group=4

musiconhold=default

context=pri-inbound

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

callerid=asreceived

useincomingcalleridonzaptransfer=yes

switchtype=national

signalling=pri_net

channel=73-95

accountcode=1714

;channels 73-95 should have accountcode 1714



Thanks!

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[asterisk-users] SIP Port 5060

[lists]

We have many clients who live in third world countries where the ISPs
purposely block traffic on port 5060.

I know we could always change the listening port in our Asterisk box.
However, doing so will affect all our other users who use port 5060 with
no problems.

Is there any other solution? I guess I could always run a second instance
of Asterisk listening on another port, but is that the cleanest and most
scalable solution?

Thanks



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Re: [asterisk-users] accountcode= placement in zapata.conf

[Andrew Joakimsen]

See the sample configfile, also its zapata.conf not zaptel.conf, the
channel=> acquires all the properties above it.

;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels which start out in a
; different context and use E & M signalling instead.
;
;context=remote
;sigalling=em
;channel => 15
;channel => 16



On 11/28/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:


Everthing should be ABOVE the channel => line for that group of channels


On 11/28/06, Damon Estep < [EMAIL PROTECTED]> wrote:

>  Can anyone tell me if accountcode= should appear before or after the
> channels definition that you want it to apply to?
>
>
>
> I have several groups of channels (PRI) defined in Zapata.conf, and wish
> to specify accountcode by group.
>
>
>
> If I put the accountcode= at the end of Zapata.conf it applies to all
> channels configured, if I put it at the end of each group section the
> behavior is inconsistent.
>
>
>
> How is accountcode parsed in Zapata.conf, does it apply to everything
> before or after the setting appears? Can it change mid-file? How?
>
>
>
> Here is the config I am working with;
>
>
>
> group=2
>
> musiconhold=default
>
> context=pri-inbound
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
> echotraining=yes
>
> callerid=asreceived
>
> useincomingcalleridonzaptransfer=yes
>
> switchtype=national
>
> signalling=pri_cpe
>
> channel=25-47
>
> accountcode=
>
> ;channels 25-47 should have accountcode=
>
>
>
> group=3
>
> musiconhold=default
>
> context=pri-inbound
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
> echotraining=yes
>
> callerid=asreceived
>
> useincomingcalleridonzaptransfer=yes
>
> switchtype=national
>
> signalling=pri_cpe
>
> channel=49-71
>
> accountcode=
>
> ;channels 49-71 should have accountcode 
>
>
>
> group=4
>
> musiconhold=default
>
> context=pri-inbound
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
> echotraining=yes
>
> callerid=asreceived
>
> useincomingcalleridonzaptransfer=yes
>
> switchtype=national
>
> signalling=pri_net
>
> channel=73-95
>
> accountcode=1714
>
> ;channels 73-95 should have accountcode 1714
>
>
>
> Thanks!
>
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>
>

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Re: [asterisk-users] SIP Port 5060

[Andrew Joakimsen]

I am wondering the same, there must be a way to CORRECTLY bind on two ports.

On 11/28/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:


We have many clients who live in third world countries where the ISPs
purposely block traffic on port 5060.

I know we could always change the listening port in our Asterisk box.
However, doing so will affect all our other users who use port 5060 with
no problems.

Is there any other solution? I guess I could always run a second instance
of Asterisk listening on another port, but is that the cleanest and most
scalable solution?

Thanks



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[asterisk-users] Best text to speech program

[Hall, Eric M.]

I'm looking to set up asterisk to call customer 3 days before the app
and remind them we will be out to see them. 
 
I'm looking for any ideas on good ways to do this. Also I think it would
be best to do some type of text to speech however I do not like the
sound of the free one . Any ideas?
 
 
Thanks!!!
 

Eric Hall


 

 
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Re: [asterisk-users] SIP Port 5060

[Patrick]

On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote:
> We have many clients who live in third world countries where the ISPs
> purposely block traffic on port 5060.
> 
> I know we could always change the listening port in our Asterisk box.
> However, doing so will affect all our other users who use port 5060 with
> no problems.
> 
> Is there any other solution? I guess I could always run a second instance
> of Asterisk listening on another port, but is that the cleanest and most
> scalable solution?

Have you tried redirecting the other port with iptables to port 5060 on
the Asterisk box?

Regards,
Patrick

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Re: [asterisk-users] Encrypted password for voicemail

[jezzzz .]

Encrypt voicemail password with Asterisk public key.
Asterisk then decrypts the password and takes the hash
of it and compares it with the hash stored in
voicemail.conf. This way the real password is never
stored in voicemail.conf and there is no way to know
what the password is just by looking at the file.

--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> On Tue, Nov 28, 2006 at 08:52:22AM -0800, je .
> wrote:
> > I was wondering if we could protect against both.
> > Sending a password encrypted would protect against
> > eavesdropping. Once the password has been
> received,
> > the hash of it is taken and compared with the hash
> of
> > the password saved, so it also takes care of a
> local
> > attacker.
> 
> Send an encypted password? Encrypted how, exactly?
> One common mistake is
> to suggest to simply send the hash, as it is
> encrypted. But this merely
> makes the hash a "password equivalent": An
> evesdroper can use the hash
> to authenticate without knowing the password.
> 
> > 
> > I could certainly use SSL/TLS, but that still
> doesn't
> > take care of a local attack to obtain the
> passwords of
> > the users.
> 
> -- 
>Tzafrir Cohen


 

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Re: [asterisk-users] When does voicemail authentication take place?

[jezzzz .]

So s,2 will only be executed if the user successfully
authenticated?

--- Luki <[EMAIL PROTECTED]> wrote:

> > Luki, thanks for the response. Could you give me
> an
> > example of the use of vmauthenticate in a very
> short
> > dialplan?
> >
> > Thanks
> > Jez
> 
> *CLI>
>   -= Info about application 'VMAuthenticate' =-
> 
> [Synopsis]
> Authenticate with Voicemail passwords
> 
> [Description]
>   VMAuthenticate([EMAIL PROTECTED]|options]):
> This application behaves the
> same way as the Authenticate application, but the
> passwords are taken from
> voicemail.conf.
>   If the mailbox is specified, only that mailbox's
> password will be considered
> valid. If the mailbox is not specified, the channel
> variable AUTH_MAILBOX will
> be set with the authenticated mailbox.
> 
> ... and ...
> 
> *CLI>
>   -= Info about application 'Authenticate' =-
> 
> [Synopsis]
> Authenticate a user
> 
> [Description]
>   Authenticate(password[|options]): This application
> asks the caller to enter a
> given password in order to continue dialplan
> execution. If the password begins
> with the '/' character, it is interpreted as a file
> which contains a list of
> valid passwords, listed 1 password per line in the
> file.
>   When using a database key, the value associated
> with the key can be anything.
> Users have three attempts to authenticate before the
> channel is hung up. If the
> passsword is invalid, the 'j' option is specified,
> and priority n+101 exists,
> dialplan execution will continnue at this location.
> 
> 
> ... so something like that (never tried it):
> 
> exten => s,1,VMAuthenticate
> exten => s,2,NoOp(Authenticated as {$AUTH_MAILBOX})
> 
> --Luki
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>
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> 



 

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Re: [asterisk-users] SIP Port 5060

[Tom Lynn]

Can you tunnel through a VPN connection?

On 11/28/06, Patrick <[EMAIL PROTECTED]> wrote:


On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote:
> We have many clients who live in third world countries where the ISPs
> purposely block traffic on port 5060.
>
> I know we could always change the listening port in our Asterisk box.
> However, doing so will affect all our other users who use port 5060 with
> no problems.
>
> Is there any other solution? I guess I could always run a second
instance
> of Asterisk listening on another port, but is that the cleanest and most
> scalable solution?

Have you tried redirecting the other port with iptables to port 5060 on
the Asterisk box?

Regards,
Patrick

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Re: [asterisk-users] Best text to speech program

[Tom Lynn]

Cepstral sounds good and it's cheap.  However, it still sounds like a
synthesized voice.

On 11/28/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:


 I'm looking to set up asterisk to call customer 3 days before the app and
remind them we will be out to see them.

I'm looking for any ideas on good ways to do this. Also I think it would
be best to do some type of text to speech however I do not like the sound of
the free one . Any ideas?


Thanks!!!


Eric Hall




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Re: [asterisk-users] Survey: In what ways do you use Asterisk at your house?

[Tom Lynn]

Earle,
I'm running Astlinux on a PIII 550 with 384 megs of ram.  Booting from a
Compact Flash card.  Non-Volatile storage on a USB Keydisk.  I have three
SIP DID numbers in three different area codes here in Western Washington via
IPKall.  I use a local SIP termination provider and also retain my Qwest
POTS line with callerid, which connects via an X100P clone board.

All in all, I've got less than $50 sunk into the system.  Past that, I added
an SPA3K, which is an utter disappointment, regardless of what old Ward
Mundy has to say about them.

I'm got various services running.  Weather reports via an cepstral speech
synthesis (runs off-server in a wmware instance).  I use it to block
unwanted callers from reaching my home based on values stored in the ASTDB
and simple dialplan logic.  Speed Dials, wakeup calls, music on hold
customized to the caller based on Caller ID, Conference bridge (remember,
I've got three DIDs via my broadband), and I'm also working on automating
retrieval of e-mail and conversion to speech so that my in-laws, who don't
have a computer, can hear their e-mails as soon as they arrive, via the
telephone, subject to common sense time of day rules.  I have DISA service
setup for the rare instance that I get an urge to call overseas from my cell
phone.

And when they build FAX capability into AstLinux, I'll use that, too.

On 11/23/06, Neil Cherry <[EMAIL PROTECTED]> wrote:


Earle Clubb wrote:

> - What service provider/technology do you use for
origination/termination?
> - What hardware/software do you use and how does it all tie together?
> - What tasks do you use * to accomplish?
> - Any other pertinent info.

Until last summer I had Asterisk doing the normal call handling
my home. You know selecting which line to call out on via an
SPA-3000 and SPA-3102. We do have trouble with the SPA's as the
echo can be quite bad or the volume is quite low (take your pick).
I'm also routing various calls to various vm-boxes and sending
selected callers to the SIT. I also had an extension that
interfaced to Mr. House home automation software. I could control
and monitor a few things in my home.

This system is no longer working due to a drive crash and the lack
of backup for parts of this setup. I'm hoping to get the time
towards the end of the year to put it back together. I may try
to integrate the voice recognition (Sphinx) into the setup also.
This was running on a 1GHz/512M/300G vanilla x86 clone. I had
printer services, DNS, DHCP, file sharing, home automation,
Asterisk and a few other things running. It's also my development
system.

--
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://www.linuxha.com/ Main site
http://linuxha.blogspot.com/My HA Blog
http://home.comcast.net/~ncherry/   Backup site
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Re: [asterisk-users] Encrypted password for voicemail

[Tzafrir Cohen]

On Tue, Nov 28, 2006 at 09:12:19PM -0800, je . wrote:
> Encrypt voicemail password with Asterisk public key.
> Asterisk then decrypts the password and takes the hash
> of it and compares it with the hash stored in
> voicemail.conf. This way the real password is never
> stored in voicemail.conf and there is no way to know
> what the password is just by looking at the file.

One minor implementation note: many phones only support digits. Thus for
them you can only use digits in the voicemail secret. All in all, the
voicemail prompt dialog may not be the best place to provide
authentication.

How exactly do you intend to interact with the user?

-- 
   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] Encrypted password for voicemail

[jezzzz .]

The secret will still only be digits but encrypted
with Asterisk's public key. For a HW phone it is a
problem (unless there's a slot for a USB or CF card),
but for soft phones it shouldn't be a problem to
obtain the server's public key. Otherwise, the key may
just be retrieved from a central repository.

> One minor implementation note: many phones only
> support digits. Thus for
> them you can only use digits in the voicemail
> secret. All in all, the
> voicemail prompt dialog may not be the best place to
> provide
> authentication.
> 
> How exactly do you intend to interact with the user?
> 
> -- 
>Tzafrir Cohen


 

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Re: [asterisk-users] vm_change_password shell?

[jezzzz .]

I'm actually only looking at the code at this point in
time (only in voicemail.c), hence my slightly
technical list of questions. How would the user find a
better way to change the password though? Isn't the
only way of changing the password by calling the
voicemail then selecting option 0 followed by option
5?

Obviously, the administrator can just as well just
change the password in voicemail.c but that's not as
convenient for the user.

Two more questions while I have your expertise...

* are then any locking problems when running
vm_change_password - voicemail.conf is opened and read
into a buffer, the password is then changed for user x
and then it's written into a new voicemail.conf.new
file. What happens if another user y wishes to change
his password at the same time as user x? Do you see
where I'm going with this?

* what's the change_password_realtime() function used
for? How does it defer from vm_change_password?

Thanks

Jez

> Use the Source, Luke.
> 
>
http://svn.digium.com/svn/asterisk/branches/1.4/apps/app_voicemail.c
> 
> In it, you'll find:
> 
> static void vm_change_password_shell(struct
> ast_vm_user *vmu, char *newpassword)
> {
>   char buf[255];
>   snprintf(buf,255,"%s %s %s
>
%s",ext_pass_cmd,vmu->context,vmu->mailbox,newpassword);
>   if (!ast_safe_system(buf))
>   ast_copy_string(vmu->password, newpassword,
> sizeof(vmu->password));
> }
> 
> 
> ext_pass_cmd is not defined in that function and
> hence a global, and its 
> value is taken from the configuration item
> "externpass".
> 
> 
> The idea is that if the user has a better way of
> changing the password,
> then the externpass script should be used to change
> the password.
> 
> -- 
>Tzafrir Cohen 


 

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