[Asterisk-Users] Linux Distribution for Asterisk server use

[TWV]

Hello,

 

My question is about which Linux distribution to choose for
Asterisk. (/me holds breath)  OK, hopefully you’re still reading,
because whatever you were thinking now, you’re thinking wrong! ;)

First of all, I want to make clear that I have read EVERY
message and reply that I could possibly find about this topic, so that includes
the dozens of messages here on the Asterisk mailinglists, on the Digum forum,
and even Google search results!

 

Still, my question was not answered!  Mainly because
the same answer always came back: “Use the one you are most comfortable
with”.  Well, I already knew that (linux is linux), but it doesn’t
apply to my situation at all!

 

Let’s make things clear and concrete now:

In my professional life, I work as a windows system &
network administrator and as a developer on the .NET platform, and have a long
and extensive experience with telecom and VoIP.  Working for a telecom
company, I now have accepted the challenge to extend our offering with
Asterisk.  During the past 6 months, I have learned as much as I possibly
could about the Asterisk PBX, successfully set up a complete test environment,
developed IVR systems, and now we feel the time is right to put these services
into production.  BUT, before we can go live, one important problem
remains: as I said, I am a Windows guy, I have a VERY profound knowledge of
Windows and manage almost 20 Windows 2003 Servers that run mission-critical
applications on a 365/24/7 basis, and support a large number of Windows
applications and Web services (some of the applications I have developed are
used by more than 25000 users every day!)  Why am I telling this?  Well,
because I want to make it clear that I am perfectly happy with my platform/OS
(windows), and have no intent whatsoever to ever change servers or application
platforms to linux (let alone my workstation).  What’s more, I have
NEVER come in to contact with linux/unix before, so I have never worked with
ANY distribution.

 

Having explained all this, it should sound logical that I
chose the AsteriskWin32 version for learning Asterisk.  Of course, I
realize that we can’t put any production system on AsteriskWin32… 
So before we can go live with Asterisk servers and services, this last issue
remains to be resolved: what Linux distribution should I choose (and learn)?

As I explained, I see Linux merely as a “necessary
evil” (because of my lack of knowledge) for running Asterisk.  So I’m
asking about the best linux distribution only to put up asterisk servers.  I’m
NOT asking for the easiest one or so (I always enjoy challenges and learning
new things), I’m asking for the best choice to build a carrier grade
telecom system, having to support thousands of users each and every day.  So
it must be reliable and easy to maintain and upgrade.  We are going to use
Asterisk in our own datacenters (supporting nation-wide services), as well as in
servers that we sell to corporations and callcenters for use as an advanced
PBX/CTI system).

So it should be clear that I’m not talking about a
hobby or home deployment here.  Our central asterisk systems for example will
have to manage DS3 or (lots of) E1 trunks…

 

If you need more information, I am happy to supply it.  I
appreciate your time and am hoping for some good suggestions and arguments
which will lead me to the correct choice for now and for the years to come.

 

Thanks!

 






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RE: [Asterisk-Users] Provider Survey

[List Receiver]

Thanks, I will look into them. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Saturday, July 02, 2005 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Provider Survey

List Receiver wrote:

> Having used Broadvoice for a while with marginal service, I want to 
> move on to another provider. So my question to the List is who is 
> good? I know now one service is perfect but somebody out there has to 
> be decent. Who have you guys had the best luck with?  
>
I suggest, if your installation is mission critial at all, you use a 
dialplan setup that has failover between two providers with failover to 
PSTN, and that you consider Teliax and NuFone as Providers. I have found 
these are serious players, they are not lemonade stands, and they offer 
failover to pstn for incoming calls also. I have never had an instance 
where both are down, but my experiene is relatively short.

-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 

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RE: RE : [Asterisk-Users] Colored asterisk -R?

[Jay Milk]

asterisk -nr

n - no colors

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, July 02, 2005 3:51 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE : [Asterisk-Users] Colored asterisk -R?
> 
> 
> Asterisk -gc
> 
> Best Regards,
> Francois BERGERET,
> France.
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part 
> de Stefan Gofferje Envoyé : samedi 2 juillet 2005 22:25 À : 
> Asterisk Users Mailing List - Non-Commercial Discussion Objet 
> : [Asterisk-Users] Colored asterisk -R?
> 
> 
> Hi folks,
> 
> when I start asterisk directly, I get a colored CLI. When 
> connect to a 
> already running asterisk with asterisk -R, it's never 
> colored, despite 
> I'm running both from the same console (tty). Is there a way to force 
> asterisk -R into color mode?
> 
> Regards,
> Stefan
> 
> -- 
>  (o_   Stefan Gofferje  | Linux Systems Specialist
>  //\   Reg'd Linux User #247167 | Network Security Specialist
>  V_/_  Heckler & Koch - the original point and click interface
> 
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RE: [Asterisk-Users] Telephoning Announcements -- Suggestions?

[Jay Milk]

That's all doable.  How many residents are you talking about? -- could
take quite a while to call them all.  Considering you have outlay in
hardware, phone-cost, utilities (a 100W computer draws $5-$10/month),
consider fixing that well as someone suggested.

> -Original Message-
> From: Scott Nelson [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, July 02, 2005 3:05 PM
> To: Asterisk Users Mailing List - Asterisk Users
> Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions?
> 
> 
> In the subdivision where I live, we have a well that time to 
> time has  
> problems.  Currently, our trustees call me, I take a message, and  
> then call the people on our "phone calling tree" and give them the  
> message.  They, in turn, pass the message on to the rest of the  
> residents.
> 
> We have a few problems with the way this works -- people 
> don't bother  
> to relay the message, or, when they do relay the message, they get  
> the message wrong.
> 
> I was thinking about using an Asterisk server -- kind of a 
> "hot line"  
> for our subdivision.  A trustee can call in and record an  
> announcement, and the server would phone each resident and play the  
> announcement to them.  At other times we can use the server to play  
> "up and coming events" and so on.
> 
> Obviously, I need to create the menus and authenticate the trustees,  
> record the message, and have that trigger an agi to create outbound  
> dial files -- and some sort of cron or agi job to get statistics of  
> who has acknowledged the message and who hasn't so that the trustee  
> can phone the resident themselves.
> 
> I'm thinking about a small PC located at one of the trustees' house  
> (or perhaps my own) with one phone line interface.  Alas, no high  
> speed internet here so I can't use VOIP instead of a POTS phone line.
> 
> Does anyone have suggestions/tips/cautions for me?  Are there any  
> applications that might help me?  Should I create a program that  
> makes the calls via the manager interface, etc.
> 
> Thanks,
> Scott Nelson
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Re: [Asterisk-Users] call forwarding, most basic case

[Yair Hakak]

hello Mike,
 we are talking about very different things here. please look at my
original mail again. I want the call recipient to be able to toggle on
and off "do not disturb". I don't want the phone to ring at all.

thanks,
 yair


On 7/3/05, Mike Hillerbrand <[EMAIL PROTECTED]> wrote:
> By user do you mean the caller (initiator of the call) or the recipient? If
> you mean that user is the call recipient, it is very easy. The caller's call
> comes to you with its Caller ID--if you want the call to go to VM, then
> don't answer the call. I use this for forwarding to other PSTN lines (cell,
> remote offices, etc..), although I would guess the same thing applies to SIP
> phones. The dial plan variables are only necessary if you want to pass
> caller ID from the originating caller through to the forwarded number. If
> you don't use the variable then the caller ID you would see would be that
> from the Asterisk configuration and not from the actual caller.
> 
> The "0" inserted into the number is helpful if you have calls forwarded
> simultaneously to your cell phone (or other) so that you can see by the zero
> that it is a forwarded call rather than a direct call to your PSTN number (I
> guess you could also use this with internal calls to distinguish calls that
> are forwarded from different extension numbers). If it is a forwarded call
> then by not answering it, it would go to Asterisk VM. If a direct call, it
> would go to whatever aswering funtion is set up on your cell phone (or other
> PSTN phone). [Please reply through the mailing list]. Mike.
> 
> -Original Message-
> From: Yair Hakak [mailto:[EMAIL PROTECTED]
> Sent: Saturday, July 02, 2005 5:05 PM
> To: Mike Hillerbrand
> Subject: Re: [Asterisk-Users] call forwarding, most basic case
> 
> 
> hi,
>  thanks for your answer, but i'm not sure i understand. this dialplan says
> 1. call the extension
> 2. set a variable with the callerIDNum
> 3. dial out to the "follow me" number with a 0 prepended to the callerID
> 4. switch the callerID back to the original
> 5. go to voicemail
> 
> how does the user turn this on and off? that's what i'm trying to do
> in my case. i want the user to be able to switch between asterisk
> calling his extension and asterisk sending the call directly to
> voicemail.
> 
> -yair
> 
> On 7/2/05, Mike Hillerbrand <[EMAIL PROTECTED]> wrote:
> > Try this
> > http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
> >
> > I used and it works well. Rather than segregate calls based on caller ID,
> it
> > carries the caller's ID through to the forwarded phone (cell phone, or
> > other?), but inserts a "0" before the number, that way you know it is an *
> > related call. If you don't answer (don't like the caller) or can't answer,
> > the call goes to voice mail.
> >
> > Mike.
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
> > Sent: Saturday, July 02, 2005 3:06 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] call forwarding, most basic case
> >
> >
> > hello all,
> >
> > i need some help and after trying the wiki i'm even more confused than i
> > was.
> >
> >  i'm trying to set up call forwarding and running into problems...
> >  i want the most basic call forwarding imaginable.
> >
> > 1. caller dials extension (say, 154)
> > 2. dialplan is updated to forward caller's extension (based on
> > CALLERIDNUM) to voicemail, instead of ringing his endpoint.
> > 3. caller is disconnected.
> >
> > as you can see, i don't want any *21 or #21, and then the number, i
> > dont even want the caller to be able to pick the number to forward to,
> > the simplest case possible, and a different extension (155) to turn
> > the forwarding off (for now, then i'll put them in a menu together or
> > something.)
> >
> > so, i know i need an extension like this:
> >
> > exten =>154,1, Answer
> > exten => 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
> > exten =>153,3, Hangup
> >
> > but line 2 is giving me fits, and the documentation is a bit thin. i'm
> > confused about the families in the database - do i have to create
> > them, or are they aready there?
> >
> > of course, if i'm barking up the wrong tree and there's a much simpler
> > way to do this please tell me.
> >
> > thanks,
> >  yair
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> >
> 
>
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[Asterisk-Users] Problem registering Asterisk Dual Server

[Joseph]

Here is my configuration everything would seems be straight forward, but
I can not register both asterisk with each other.

Both asterisks have Static IP but they are  behind firewall/router, so
it means I have to use Register statement.  
I'm a bit confused about the register statement.  
How can they can register with each other when both firewalls are
blocking port 4569?

Do I have to open ports 4569 on both servers in order to register them
with each other?


SERVER 1:
register => Ast-1-in:[EMAIL PROTECTED]

[Ast-2-in]
type=user
secret=password2
context=incoming
disallow=all
allow=ulaw
;trunk=yes

[Ast-2-out]
type=peer
disallow=all
allow=ulaw
host=dynamic

==
SERVER 2:
register => Ast-2-in:[EMAIL PROTECTED]

[Ast-1-in]
type=user
secret=password1
context=incoming
disallow=all
allow=ulaw
;trunk=yes

[Ast-1-out]
type=peer
disallow=all
allow=ulaw
host=dynamic


-- 
#Joseph
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Re: [Asterisk-Users] IAX DTMF Problem...

[Mark Edwards]

 
I hear you. "background" is in definitely in use in my extensions.conf here. 
Hopefully this partially accounts for the 10% of times when it _does_ work! ;-)Mark
 
On 7/3/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Saturday 02 July 2005 19:56, Mark Edwards wrote:> Thanks guys - appreciate the comments. I understand that IAX does not
> support inband dtmf, but I still can't fathom why 9 times out of 10 my *> box is ignoring DTMF's even though they are showing up in the IAX2 protocol> debug output. The really annoying thing is that I can't consistently
> reproduce the issue - every now and again it just works. I have logged a> bug 004631 and will see what happens from here.You aren't by any chance trying to pick up DTMF while Playback() is executing
are you?  Use Background() for that.-A.___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917
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Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)

[Ronald Wiplinger]

Robert Goodyear wrote:




On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:


I am confused about one of my installed server

The dial plan seems to be ok, but sometimes NOTHING happens if I try 
to dial an extension (from X-Lite), next time it is done.


X-Lite does not have a tone, nothing and does also have no time out. 
It seems it is not connected to the server. However, a sip show users 
/ sip show peers   shows that the phone is connected.




SIP clients generate their own dialtone, so if you've got no tone, 
that sounds suspicious of a problem with the client itself. I assume 
you've debugged the problem by registering a hard SIP client on that 
server?




The CLI prompt does not show anything either. It is like the phone is 
not talking to asterisk at all.

sip show users/peers   does show the phone.


bye

Ronald

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[Asterisk-Users] Festival long starting time

[Ronald Wiplinger]

I have installed festival and it works, but it takes a long time, till 
it starts. People hang up before they can listen the info.


Q:

1. How can I shorten this time?
2. If two people call this extension, the second one must wait till the 
first one is finished. Is there a way to serve multiple poeple?
3. I have setup festival as explained in the first example. What are the 
advantages / disadvantages of the three methodes?

4. How long can a info be?
5. Besides weather.agi, what other application exist or what could be 
useful to use festival within asterisk



bye

Ronald

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Re: [Asterisk-Users] hidecallerid on analog line

[Chris Travers]

chawki hammoud wrote:


In the ISDN case, setcallerid or hidecallerid can be
configured and I am aware that  Asterisk doesn't
support that on analog line. My question is whethere 
there is something like add-on script or hardware that

will do the job. The teleco company provide the
callerid service, but no private number service.
 

If I understand the signalling architecture correctly, if your telco 
does not provide the service, you are out of luck.


A quick explenation:  ISDN (PRI for the purpose of this example) uses 
out of band signalling to provide what is essentially a trunk line plus 
a signalling channel to your PBX.  Basically one of the DS0 channels on 
the T1 is used for signalling and called the D channel (while your voice 
DS0's are called B channels).


Analog lines are completely different.  In general, very little if any 
signalling is expected to be carried over an analog line.  Therefore, 
the advanced signalling capabilities that ISDN offer are unavailable on 
analog simply because the telco architecture doesn't support it.  
Basically, in this case, *all* caller-id data originates from the 
telco's switch.  The only way to suppress this information is to tell 
the switch to suppress it.  Unless they offer such a service (prefixing 
the number with something, for example, to activate it), you have no 
access to the system.  In this case there is nothing you can add to make 
this work.


Best Wishes,
Chris Travers
Metatron Technology Consulting


--- Robert Webb <[EMAIL PROTECTED]> wrote:

 


On Wed, 29 Jun 2005 13:56:00 -0700 (PDT)
 chawki hammoud <[EMAIL PROTECTED]> wrote:
   


Is there a way to hide the callerid on analog line
 


on
   


outgoing calls. Any ideas whether it could be done
through configuration, a script or hardware.

Thanks;

 


It would have to be done through who ever provides
your 
POTS service. They provide the caller ID to who you
are 
calling. Some have the option to block it. Asterisk
cannot 
be configured to do this.

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Re: [Asterisk-Users] IAX DTMF Problem...

[Andrew Kohlsmith]

On Saturday 02 July 2005 19:56, Mark Edwards wrote:
> Thanks guys - appreciate the comments. I understand that IAX does not
> support inband dtmf, but I still can't fathom why 9 times out of 10 my *
> box is ignoring DTMF's even though they are showing up in the IAX2 protocol
> debug output. The really annoying thing is that I can't consistently
> reproduce the issue - every now and again it just works. I have logged a
> bug 004631 and will see what happens from here.

You aren't by any chance trying to pick up DTMF while Playback() is executing 
are you?  Use Background() for that.

-A.
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[Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head

[Keith Caldwell]

Ok, after hours of research I finally found the problem. I found a  
document from digium at


http://www.digium.com/asterisk_handbook/zapata.conf.html

 which states that everything above the channel=x statement applies  
to that interface which seems a little backwards to me. After  
reconfiguring I have


context=internal
signalling=fxo_ks
callerid="Keith" <100>
channel=1

context=pstn-in
signalling=fxs_ks
callerid=asrecieved
channel=4

Just in case anyone else has the same problem.

Keith


On Jul 1, 2005, at 10:35 PM, Keith Caldwell wrote:

I just downloaded the latest asterisk and zaptel from cvs and I'm  
having trouble setting the callerid on the fxs interface. From what  
I've read you should be able to set it in the zapata.conf but I  
can't get it to work. I had a cheap X100P clone but recently bought  
the Dev Kit PCI .  I've built a seperate server and have everything  
up and running except everytime I go to check my voicemail I have  
to enter my extension each time. Here is the exten for voicemail  
that I have set up.


[vmail-main]
exten => 580,1,Wait,1
exten => 580,2,Answer
exten => 580,3,VoiceMailMain([EMAIL PROTECTED])
exten => 580,4,Wait,2
exten => 580,5,hangup



I've set up a test extension that will say cid
exten => 222,1,Wait,2
exten => 222,2,SayDigits(${CALLERIDNUM})
exten => 222,3,Hangup

It works with a sip soft phone I've tested it with.


Here is the zapata.conf

[channels]

busydetect=no
musiconhold=default
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes

immediate=no

context=internal
signalling=fxo_ks
channel=1

context=pstn-in
callerid="Keith" <100>

signalling=fxs_ks
channel=4

callerid=asrecieved

Any Idea what I'm doing wrong?


Keith




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[Asterisk-Users] Connecting * to a Ericsson BP250

[Christian Keiser]

Hi List, 

This is somewhat off-topic since the problem itself isn’t asterisk but the
Ericsson BP250 I want to connect to. 
But since there have been a couple of posts relating in part to that system
I am hoping someone can help me out. 

What we want to do:

PRI <---> BP250 <---> Asterisk

Currently the BP250 handles extension 000 - 899. 
We now would like to forward a chunk of 100 extensions to the asterisk
system. We would like to use asterisk to deploy voice-mail fax services and
voip/SIP connections to home-office users.
Also we would like to connect to different location using IAX.

The Problem: 

The BP250 was installed and is serviced by an Ericsson vendor. Since we have
a service contract I called them and explained what I needed -> "a link to
my internal asterisk system forwarding those 100 extensions"
The tech guy then wanted to know what I wanted it for. SO I explained that
we wanted to deploy voice-mail and connect home users using SIP as well as
IAX peerings to other locations. 

He said "It's not possible" But that he would talk to a colleague to find a
solution and that he would get back to me. 
The following week I received and offer for 1. A voice-mail add-on für our
BP250 2. A Voip interface for the BP250 alternatively they would be happy to
discuss a new system based on VOIP entirely. 

Since I couldn't believe that there was no solution to my Problem I asked
for the BP250 / RASC Documentation. According to our vendor such User
Manuals do not exist and that knowledge of the system is only available
through Ericsson training courses. 

So I am somewhat stuck in my efforts. I googled for the past week but
couldn't find anything that would really help me. 

I still don't believe that the BP250 is incapable of forwarding those 100
extensions to an internal PRI. 

Does anyone know if it is possible?? And maybe even how ? 

If I knew how I could exert some pressure on my vendor to implement it.
Otherwise I would have to take his word and my company would have to
purchase the addons for the BP250

Thanks in advance to all of you. 


Best regards

Christian 





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[Asterisk-Users] Audiocodes MP-108 FXO to Asterisk HELP

[Darren Wright]

 
Does anyone have configs on the MP-108 FXO to asterisk setup?  I'm doing
my best with the limited docs, but having very little success.

Thanks,

-Darren
 
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Re: [Asterisk-Users] Problem with DTFM and complex international setup

[Mark Edwards]

Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631?
 
Mark 
On 7/3/05, Mohit Muthanna <[EMAIL PROTECTED]> wrote:
Right... that's the one. My mistake.On 7/1/05, Rob Scott <[EMAIL PROTECTED]
> wrote:> I don't find this option in the Makefile.> I find RADIO_RELAX which is something to do with radios and DTMF.>> -Original Message-> From: 
[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of Mohit> Muthanna> Sent: 01 July 2005 23:24
> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [Asterisk-Users] Problem with DTFM and complex> international setup>> Try compiling Asterisk with RELAX_DTMF (See Makefile).
>> Mohit.>> On 7/1/05, Rob Scott <[EMAIL PROTECTED]> wrote:> > We have some guys working in the US who can't always dial back to our
> > company in Europe easily (lots of clients require authorization to> > make international calls), so I set up the following:> >> >- ipkall.com number links to a FWD number
> >- office Asterisk box registers with FWD> >> > Then I programmed Asterisk to accept office extension number using> > DTFM tones.> > This works OK.> >> > Then I programmed Asterisk so that it is possible, using a PIN code,
> > to dial out from Asterisk onto the local PSTN.> >> > This also works occasionally.> > Looking at the message from the Asterisk box it is clear that> > sometimes numbers are missed or repeated in the dial string. This I
> > suspect is because Asterisk is listening to the DTMF tones but the> > signal is dropped; sometimes the drop means that a whole digit is> > dropped and sometimes is means that a digit is repeated.
> >> > Does anyone know how I can fix this to make it more reliable> > (out-of-band DTMF?) or a better way to achieve a reliable setup?> > ___
> > Asterisk-Users mailing list> > Asterisk-Users@lists.digium.com> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:> >http://lists.digium.com/mailman/listinfo/asterisk-users> >>
>> --> Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of> people. Those who understand binary, and those who don't."> ___
> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>--Mohit Muthanna [mohit (at) muthanna (uhuh) com]
"There are 10 types of people. Those who understand binary, and thosewho don't."___Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917
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[Asterisk-Users] HW Capacity plan - How many Digium is recomended per server

[Manuel Soto]

Hello all,

I'm evaluating a VRU project which has huge requirements. I'm looking
for metrics but I haven't found anything that cover my requirements

Initial estimation:

Erlang  61.450
BTH 25.980
T1 req. 88

Digium HW support 4 T1 per card, assuming that each server host 3
Wildcard then 7 clustered servers will be required to support this load,
but, How many Digium cards is recommended per server?, servers may be 2
CPU Xeon 2.4GHz to 3GHz

Thanks,
MS
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Re: [Asterisk-Users] IAX DTMF Problem...

[Mark Edwards]

Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce the issue - every now and again it just works. I have logged a bug 004631 and will see what happens from here.

 
cheers,
 
Mark 
On 7/3/05, Mohit Muthanna <[EMAIL PROTECTED]> wrote:
>  From what I understand, that is one of the reasons with SIP inband> doesn't mix well with any codec other than 
G.711.I believe it's just the ulaw/alaw PCM codecs that allow inband DTMFfor SIP. Anything else will just chew it up.Mohit.___Asterisk-Users mailing list
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Mark P. EdwardsFWD: 667917
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[Asterisk-Users] LDAP search application for Asterisk

[Juan Jose Comellas]

I'm sending an Asterisk module I've written to see if anybody finds it useful 
or wants to provide some feedback. The name of the module is app_ldap and the 
application it provides is named LDAPSearch.

LDAPSearch allows any kind of searches on an LDAP directory from the Asterisk 
dialplan. It returns its results using channel variables and it was 
specifically designed to be able to dial by name (users, contacts from an 
addressbook, etc.). One of its most interesting features is the extension it 
adds to the LDAP filter syntax (#= operator) to be able to match LDAP entries 
against a series of digits using the standard touch tone phone key mapping to 
make its comparisons.

The syntax used by the application is the following:

LDAPSearch(filter,[attr_1[:attr_2[:...]]][,max entries[,sort attr[,base 
DN[,scope)

The documentation explaining its configuration and usage is included in 
doc/html/app_ldap.html.

The only strange thing needed to build it is SCons (http://www.scons.org), 
which btw is an excellent build tool. You'll also need an LDAP client library 
and an LDAP server to test it. Everything has been tested using the OpenLDAP 
client library against both the OpenLDAP server and Microsoft Active 
Directory 2000.

You can download it from:

http://comellas.com.ar/app_ldap/app_ldap-0.1.0.tar.bz2

You'll need some knowledge of LDAP to be able to use this application 
successfully.

Please report any problems you may have with it.

-- 
Juan Jose Comellas
([EMAIL PROTECTED])
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Re: [Asterisk-Users] What to use h323 or oh323 ???

[Isamar Maia]

It's a little bit hard to compile but Try oh323 first.

Although, There will be some few situations that H323 will work better
than oh323. So, have both.

Isamar


On Sat, 2 Jul 2005, Adeel -31 wrote:

> I m new to asterisk n i've got an IP phone that supports h323 protocol 
> but i dont know how to configure asterisk to use it... i m comfortable in 
> using sip & iax softphones but there is no h323.conf in /etc/asterisk/   
>  i read that i've to compile some files but i m confused regarding h323 & 
> oh323  .. which one should i use.. plz tell me or atleast give some 
> helpful link
>
> __
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com

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Re: [Asterisk-Users] IAX DTMF Problem...

[Mohit Muthanna]

>  From what I understand, that is one of the reasons with SIP inband
> doesn't mix well with any codec other than G.711.

I believe it's just the ulaw/alaw PCM codecs that allow inband DTMF
for SIP. Anything else will just chew it up.

Mohit.
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Re: [Asterisk-Users] Problem with DTFM and complex international setup

[Mohit Muthanna]

Right... that's the one. My mistake.

On 7/1/05, Rob Scott <[EMAIL PROTECTED]> wrote:
> I don't find this option in the Makefile.
> I find RADIO_RELAX which is something to do with radios and DTMF.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mohit
> Muthanna
> Sent: 01 July 2005 23:24
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Problem with DTFM and complex
> international setup
> 
> Try compiling Asterisk with RELAX_DTMF (See Makefile).
> 
> Mohit.
> 
> On 7/1/05, Rob Scott <[EMAIL PROTECTED]> wrote:
> > We have some guys working in the US who can't always dial back to our
> > company in Europe easily (lots of clients require authorization to
> > make international calls), so I set up the following:
> >
> >- ipkall.com number links to a FWD number
> >- office Asterisk box registers with FWD
> >
> > Then I programmed Asterisk to accept office extension number using
> > DTFM tones.
> > This works OK.
> >
> > Then I programmed Asterisk so that it is possible, using a PIN code,
> > to dial out from Asterisk onto the local PSTN.
> >
> > This also works occasionally.
> > Looking at the message from the Asterisk box it is clear that
> > sometimes numbers are missed or repeated in the dial string. This I
> > suspect is because Asterisk is listening to the DTMF tones but the
> > signal is dropped; sometimes the drop means that a whole digit is
> > dropped and sometimes is means that a digit is repeated.
> >
> > Does anyone know how I can fix this to make it more reliable
> > (out-of-band DTMF?) or a better way to achieve a reliable setup?
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> --
> Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of
> people. Those who understand binary, and those who don't."
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
Mohit Muthanna [mohit (at) muthanna (uhuh) com]
"There are 10 types of people. Those who understand binary, and those
who don't."
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Re: [Asterisk-Users] call forwarding, most basic case

[Ferdy Riphagen]

Yair,


One option is like this:

1) User dials ext. 154 to activate call forward (to voicemail)
2) User dials ext. 155 to de-activate call forward
3) Macro to check incoming calls for database entry's
4) The local extention must use that macro (or other way of screening)

1)
exten => 154,1,Answer
exten => 154,2,Set(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
exten => 154,3,Hangup

2)
exten => 155,1,Answer
exten => 155,2,DBdel(CFIM/${CALLERIDNUM})
exten => 155,3,Hangup

3)
[macro-test]
exten => s,1,Set(CFIM=${DB(CFIM/${ARG1})})
exten => s,2,GotoIf(${CFIM} = "CFIM/${ARG1}"?1|1)
exten => s,3,Dial(${ARG2}|${ARG3}|${ARG5})

exten => 1,1,VoiceMail(u${CFIM})

4)
exten => 202,1,Macro(test|${EXTEN}|SIP/202|15||tr)


Regards,

/* Ferdy */

http://asterisk.nsec.nl
info(AT)nsec(DOT)nl



Yair Hakak wrote:
> hello all,
> 
> i need some help and after trying the wiki i'm even more confused than i was.
> 
>  i'm trying to set up call forwarding and running into problems...
>  i want the most basic call forwarding imaginable.
> 
> 1. caller dials extension (say, 154)
> 2. dialplan is updated to forward caller's extension (based on
> CALLERIDNUM) to voicemail, instead of ringing his endpoint.
> 3. caller is disconnected.
> 
> as you can see, i don't want any *21 or #21, and then the number, i
> dont even want the caller to be able to pick the number to forward to,
> the simplest case possible, and a different extension (155) to turn
> the forwarding off (for now, then i'll put them in a menu together or
> something.)
> 
> so, i know i need an extension like this:
> 
> exten =>154,1, Answer 
> exten => 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM}) 
> exten =>153,3, Hangup
> 
> but line 2 is giving me fits, and the documentation is a bit thin. i'm
> confused about the families in the database - do i have to create
> them, or are they aready there?
> 
> of course, if i'm barking up the wrong tree and there's a much simpler
> way to do this please tell me.
> 
> thanks,
>  yair
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RE: [Asterisk-Users] call forwarding, most basic case

[Mike Hillerbrand]

By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines (cell,
remote offices, etc..), although I would guess the same thing applies to SIP
phones. The dial plan variables are only necessary if you want to pass
caller ID from the originating caller through to the forwarded number. If
you don't use the variable then the caller ID you would see would be that
from the Asterisk configuration and not from the actual caller.

The "0" inserted into the number is helpful if you have calls forwarded
simultaneously to your cell phone (or other) so that you can see by the zero
that it is a forwarded call rather than a direct call to your PSTN number (I
guess you could also use this with internal calls to distinguish calls that
are forwarded from different extension numbers). If it is a forwarded call
then by not answering it, it would go to Asterisk VM. If a direct call, it
would go to whatever aswering funtion is set up on your cell phone (or other
PSTN phone). [Please reply through the mailing list]. Mike.

-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case


hi,
 thanks for your answer, but i'm not sure i understand. this dialplan says
1. call the extension
2. set a variable with the callerIDNum
3. dial out to the "follow me" number with a 0 prepended to the callerID
4. switch the callerID back to the original
5. go to voicemail

how does the user turn this on and off? that's what i'm trying to do
in my case. i want the user to be able to switch between asterisk
calling his extension and asterisk sending the call directly to
voicemail.

-yair

On 7/2/05, Mike Hillerbrand <[EMAIL PROTECTED]> wrote:
> Try this
> http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
>
> I used and it works well. Rather than segregate calls based on caller ID,
it
> carries the caller's ID through to the forwarded phone (cell phone, or
> other?), but inserts a "0" before the number, that way you know it is an *
> related call. If you don't answer (don't like the caller) or can't answer,
> the call goes to voice mail.
>
> Mike.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
> Sent: Saturday, July 02, 2005 3:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] call forwarding, most basic case
>
>
> hello all,
>
> i need some help and after trying the wiki i'm even more confused than i
> was.
>
>  i'm trying to set up call forwarding and running into problems...
>  i want the most basic call forwarding imaginable.
>
> 1. caller dials extension (say, 154)
> 2. dialplan is updated to forward caller's extension (based on
> CALLERIDNUM) to voicemail, instead of ringing his endpoint.
> 3. caller is disconnected.
>
> as you can see, i don't want any *21 or #21, and then the number, i
> dont even want the caller to be able to pick the number to forward to,
> the simplest case possible, and a different extension (155) to turn
> the forwarding off (for now, then i'll put them in a menu together or
> something.)
>
> so, i know i need an extension like this:
>
> exten =>154,1, Answer
> exten => 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
> exten =>153,3, Hangup
>
> but line 2 is giving me fits, and the documentation is a bit thin. i'm
> confused about the families in the database - do i have to create
> them, or are they aready there?
>
> of course, if i'm barking up the wrong tree and there's a much simpler
> way to do this please tell me.
>
> thanks,
>  yair
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>

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RE: [Asterisk-Users] Sipura SPA2000 behind NAT

[Thierry Wehr]

Hello

This iptables setup won't work
You need specific rules for the incoming UDP packets with status ESTABLISHED
and RELATED like these simple ones
Remember it's a statefull firewall.

In the nat section

-A POSTROUTING -p udp -m udp -m state --state RELATED -j MASQUERADE
-A POSTROUTING -p udp -m udp -m state --state ESTABLISHED -j MASQUERADE


And in the filter section

-A FORWARD -p udp -m udp -m state --state RELATED -j ACCEPT
-A FORWARD -p udp -m udp -m state --state ESTABLISHED -j ACCEPT

Best regards
Thierry

> -Message d'origine-
> De : [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] De la part 
> de Guillermo Salas M
> Envoyé : samedi 2 juillet 2005 22:56
> À : asterisk-users@lists.digium.com
> Objet : RE: [Asterisk-Users] Sipura SPA2000 behind NAT
> 
> Carlos,
> 
> Thank you for your fast response :) , this is the output of 
> iptables -nL on my linux box:
> 
> [EMAIL PROTECTED]:/home/guillermo # iptables -nL Chain INPUT 
> (policy ACCEPT)
> target prot opt source   destination
> 
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
> ACCEPT all  --  192.168.0.0/24   0.0.0.0/0
> ACCEPT all  --  0.0.0.0/0192.168.0.0/24
> 
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
> 
> [EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat Chain 
> PREROUTING (policy ACCEPT)
> target prot opt source   destination
> 
> Chain POSTROUTING (policy ACCEPT)
> target prot opt source   destination
> MASQUERADE  all  --  192.168.0.0/24   0.0.0.0/0
> 
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
> 
> 
> This is my very-small and simple firewall script:
> [EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall # 
> Cargar Modulos modprobe ip_tables modprobe ip_nat_ftp 
> modprobe ip_conntrack_ftp modprobe ip_nat_irc modprobe 
> ip_conntrack_irc
> 
> # Habilitar el forward
> echo 1 > /proc/sys/net/ipv4/ip_forward
> 
> # Flush
> iptables -X
> iptables -F
> iptables -X -t nat
> iptables -F -t nat
> 
> # Habilitar nat para 192.168.0.0/24
> iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j 
> MASQUERADE # Permitir el forward para 192.168.0.0/24 iptables 
> -A FORWARD -s 192.168.0.0/24 -j ACCEPT iptables -A FORWARD -d 
> 192.168.0.0/24 -j ACCEPT
> 
> # EOF
> 
> 
> On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote:
> > Guillermo,
> > 
> > This is an issue with your router. Do you have open the 
> ports 5060 for SIP?
> > Also, RTP needs to be open from 16384 to 32767.
> > 
> > Saludos,
> > 
> > Carlos Alperin
> > Senior System Engineer
> > Seneca Communications, LLC
> > [EMAIL PROTECTED]
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Guillermo Salas M
> > Sent: Saturday, July 02, 2005 4:13 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Sipura SPA2000 behind NAT
> > 
> > Hi, I've one Sipura SPA2000 at home behind a linuxbox with 
> two network 
> > adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
> > 
> > 
> > ___ HOME ___   OFFICE 
> > SPA2000 <---> Linux Box   <--> Asterisk Box
> > 192.168.0.253192.168.0.1 eth1  200.93.xxx.a
> >  200.93.xxx.b eth0
> > 
> > My problem is when I try to call to any trunk or extention 
> I can the 
> > audio when the destination is ringing, but I can hear the 
> voice of the 
> > person when it reponds. The person in the other side can 
> hear me, but 
> > I can not hear anything from him. I can not hear the voice 
> prompts for 
> > the voicemail (*98) or the operator voice, but can leave voice 
> > messages to other SIP devices and they can hear my messages.
> > 
> > This is my sip.conf
> > [105]
> > username=105
> > type=friend
> > secret=105
> > qualify=no
> > port=5060
> > nat=yes
> > [EMAIL PROTECTED]
> > host=dynamic
> > dtmfmode=rfc2833
> > context=from-internal
> > canreinvite=no
> > callerid="Guilllermo Salas HOME" <105>
> > 
> > My ext on line 1 of the Sipura is 105, and is registred 
> with the * box:
> > -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600
> > 
> > asterisk*CLI> sip show peer 105
> > asterisk*CLI>
> > 
> >   * Name   : 105
> >   Secret   : 
> >   MD5Secret: 
> >   Context  : from-internal
> >   Language : es
> >   FromUser :
> >   FromDomain   :
> >   Callgroup:  (0)
> >   Pickupgroup  :  (0)
> >   Mailbox  : [EMAIL PROTECTED]
> >   LastMsgsSent : 2
> >   Dynamic  : Yes
> >   Expire   : 4
> >   Expiry   : 900
> >   Insecure : No
> >   Nat  : Always
> >   ACL  : No
> >   CanReinvite  : No
> >   PromiscRedir : No
> >   DTMFmode : rfc2833
> >   LastMsg  : 0
> >   ToHost   :
> >   Addr->IP : 200.93.xxx.xb Port 5060
> >   Defaddr->IP  : 0.0.0.0 Port 5060
> >   Username : 1

RE: [Asterisk-Users] play message to callee before connect toinco mingcall

[mattf]

You can send both paties to a meetme conference with Manager Redirect. Or if
you are feeling more adventurous you could load the Manager Bridge patch
that I posted to the bugtracker two months ago. It allows bridging of any
two existing channels together through a manager action:
http://bugs.digium.com/view.php?id=4297

MATT---

-Original Message-
From: Roland Zagler [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] play message to callee before connect
toincomingcall


sorry for the misunderstanding, robert!

the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should be
connected
to the sip phone 100.

the behaviour is just like MoH, but the problem is, that the caller has
to hear a
soundfile from the beginning, and MoH does not do this as the sound
files inside
the directory specified in musiconhold.conf are started when asterisk is
started.

an idea i had was to write an agi script which is called in dialplan
before i issue a
playback command to the caller, which is producing a callfile to call
sip phone 100,
but how can i "connect" these 2 calls afterwards? is there a possibility
to connect
the answered sip phone 100 to the incoming call (that is still listening
to the playback
command)?

roland

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Saturday, July 02, 2005 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] play message to callee before connect
toincomingcall


On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:

> Thank you, Robert!
>
> The announcementfile plays well, but at Dial-option "m" i have to
> specify a MoH class,
> that is something i cannot use (as i wrote in my post).

Noted, which is why I offered option two.

>
> Background command waits for a user input, but the caller should be
> connected to
> SIP Phone 100 after it has answered and the announcement has been
> played.

But it doesn't REQUIRE input. Background completes when then sound file 
ends. Are you saying you want to move on to announcing the call to the 
callee as soon as it comes in while the caller is listening to the 
soundfile?

I was following your sequential steps in your post, but if you intend 
to fork the process and be doing two things at once, then it's more 
complex.


> Before
> connecting to SIP Phone 100 the caller should hear a soundfile...
>
> wiki says nothing about an Dial-option to play a soundfile to the 
> caller
> ;-(

Sure it does... BACKGROUND.

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Re: [Asterisk-Users] Colored asterisk -R?

[C. Hatton Humphrey]

> I'm running both from the same console (tty). Is there a way to force
> asterisk -R into color mode?

It works fine for me to run astersik -rc

Hatton
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Re: [Asterisk-Users] Colored asterisk -R?

[Stefan Gofferje]

[EMAIL PROTECTED] schrieb:


Asterisk -gc
 


I don't see a -R in that...

Regards,
Stefan

--
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//\   Reg'd Linux User #247167 | Network Security Specialist
V_/_  Heckler & Koch - the original point and click interface

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[Asterisk-Users] Re: passing through MWI info from SBC

[Mike Myers]

Jon Radon wrote:

>Woah woah woah.. why not just disable SBC voicemail
>and have asterisk
>handle it?  I don't understand why you would go to
>such great lengths
>when you can just have Asterisk deal with it.


Jon, etc..., the issue here is her family all uses
special features of SBC voicemail.  E.g. Her mom
leaves a VM for my wife at 10:30 PM after the baby is
asleep without fear of ringing the phones and possibly
waking up the baby.  They use it like email.  They
sign in and create messages, and even send a message
to multiple recipients, all without ever ringing the
phone.  When a VM is recieved the destination party's
MWI lights and whenever this is detected, they dail
the SBC VM and listen to the message, reply, etc...

Since SBC VM doesn't network with Asterisk or anyone
else, I need to find a way of passing the MWI info
from the SBC line to the handsets in the house, else
my wife won't know the she has VM from her family, and
then the whole system breaks down.  

As I said before, everyone in her family has SBC VM. 
Anytime one of the husbands moved someone off of SBC,
they were back in less than a week after realizing it
broke how the family used voicemail.  

I can't get my wife off of SBC VM and onto Asterisk
without moving everyone else off, which is very hard,
and then I'd have to do training and answer any
complaints, etc...  

So that's why I need to be able to have asterisk
detect  a message waiting on the SBC VM system and
display it.  Unless I can be backward compatible with
the current analog phone system we have, the whole
asterisk project will be nixed, and I'll have to go
buy some more Venture handsets to replace the broken
ones.  :-(

Thanks,
Mike


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RE: [Asterisk-Users] Sipura SPA2000 behind NAT

[Carlos Alperin]

Guillermo,

I'm not very expert with iptables, but this is the issue:

I don't see the forwarding from the ip of the sipura box (& that should be
the only one to receive both UDP & RTP traffic on the 5060 & 16384 to 32767
ports. On the other hand, the Asterisk box is also in an fix ip, so the
traffic on UDP and the RTP only needs to be open between this two IP's.

If you don't get audio, it is because UDP packets are drop when they come
back to you from Asterisk, or from the other side of the call.

What I see is that your firewall looks to be open on all IP's, but doesn't 
Mean that is in the right ports.

Sorry if my advice is difuse, but as I said before, I'm not an expert on
iptables.

You can do an easy test, if you have an cheap Router as a Linksys or D-Link
You only need to forward the ports to the ip address of the Sipura box.

Is all that you need to make this work. 

Lamento no poder ser mas especifico.

Saludos

Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M
Sent: Saturday, July 02, 2005 4:56 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Sipura SPA2000 behind NAT

Carlos,

Thank you for your fast response :) , this is the output of iptables -nL
on my linux box:

[EMAIL PROTECTED]:/home/guillermo # iptables -nL
Chain INPUT (policy ACCEPT)
target prot opt source   destination

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
ACCEPT all  --  192.168.0.0/24   0.0.0.0/0
ACCEPT all  --  0.0.0.0/0192.168.0.0/24

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination

[EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat
Chain PREROUTING (policy ACCEPT)
target prot opt source   destination

Chain POSTROUTING (policy ACCEPT)
target prot opt source   destination
MASQUERADE  all  --  192.168.0.0/24   0.0.0.0/0

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination


This is my very-small and simple firewall script:
[EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall
# Cargar Modulos
modprobe ip_tables
modprobe ip_nat_ftp
modprobe ip_conntrack_ftp
modprobe ip_nat_irc
modprobe ip_conntrack_irc

# Habilitar el forward
echo 1 > /proc/sys/net/ipv4/ip_forward

# Flush
iptables -X
iptables -F
iptables -X -t nat
iptables -F -t nat

# Habilitar nat para 192.168.0.0/24
iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j MASQUERADE
# Permitir el forward para 192.168.0.0/24
iptables -A FORWARD -s 192.168.0.0/24 -j ACCEPT
iptables -A FORWARD -d 192.168.0.0/24 -j ACCEPT

# EOF


On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote:
> Guillermo,
> 
> This is an issue with your router. Do you have open the ports 5060 for
SIP?
> Also, RTP needs to be open from 16384 to 32767.
> 
> Saludos,
> 
> Carlos Alperin
> Senior System Engineer 
> Seneca Communications, LLC
> [EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
> Salas M
> Sent: Saturday, July 02, 2005 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sipura SPA2000 behind NAT
> 
> Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
> adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
> 
> 
> ___ HOME ___   OFFICE 
> SPA2000 <---> Linux Box   <--> Asterisk Box
> 192.168.0.253192.168.0.1 eth1  200.93.xxx.a
>  200.93.xxx.b eth0
> 
> My problem is when I try to call to any trunk or extention I can the
> audio when the destination is ringing, but I can hear the voice of the
> person when it reponds. The person in the other side can hear me, but I
> can not hear anything from him. I can not hear the voice prompts for the
> voicemail (*98) or the operator voice, but can leave voice messages to
> other SIP devices and they can hear my messages.
> 
> This is my sip.conf
> [105]
> username=105
> type=friend
> secret=105
> qualify=no
> port=5060
> nat=yes
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> context=from-internal
> canreinvite=no
> callerid="Guilllermo Salas HOME" <105>
> 
> My ext on line 1 of the Sipura is 105, and is registred with the * box:
> -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600
> 
> asterisk*CLI> sip show peer 105
> asterisk*CLI>
> 
>   * Name   : 105
>   Secret   : 
>   MD5Secret: 
>   Context  : from-internal
>   Language : es
>   FromUser :
>   FromDomain   :
>   Callgroup:  (0)
>   Pickupgroup  :  (0)
>   Mailbox  : [EMAIL PROTECTED]
>   LastMsgsSent : 2
>   Dynamic  : Yes
>   Expire   : 4
>   Expiry   : 900
>   Insecure : No
>   Nat  : Always
>   ACL  : No
>   CanReinvite  : No
>   PromiscRedir : No
>   DTMFmode : rfc2833
>   LastMsg  : 0
>   ToHost   :
>   Ad

RE: [Asterisk-Users] Sipura SPA2000 behind NAT

[Guillermo Salas M]

Carlos,

Thank you for your fast response :) , this is the output of iptables -nL
on my linux box:

[EMAIL PROTECTED]:/home/guillermo # iptables -nL
Chain INPUT (policy ACCEPT)
target prot opt source   destination

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
ACCEPT all  --  192.168.0.0/24   0.0.0.0/0
ACCEPT all  --  0.0.0.0/0192.168.0.0/24

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination

[EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat
Chain PREROUTING (policy ACCEPT)
target prot opt source   destination

Chain POSTROUTING (policy ACCEPT)
target prot opt source   destination
MASQUERADE  all  --  192.168.0.0/24   0.0.0.0/0

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination


This is my very-small and simple firewall script:
[EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall
# Cargar Modulos
modprobe ip_tables
modprobe ip_nat_ftp
modprobe ip_conntrack_ftp
modprobe ip_nat_irc
modprobe ip_conntrack_irc

# Habilitar el forward
echo 1 > /proc/sys/net/ipv4/ip_forward

# Flush
iptables -X
iptables -F
iptables -X -t nat
iptables -F -t nat

# Habilitar nat para 192.168.0.0/24
iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j MASQUERADE
# Permitir el forward para 192.168.0.0/24
iptables -A FORWARD -s 192.168.0.0/24 -j ACCEPT
iptables -A FORWARD -d 192.168.0.0/24 -j ACCEPT

# EOF


On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote:
> Guillermo,
> 
> This is an issue with your router. Do you have open the ports 5060 for SIP?
> Also, RTP needs to be open from 16384 to 32767.
> 
> Saludos,
> 
> Carlos Alperin
> Senior System Engineer 
> Seneca Communications, LLC
> [EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
> Salas M
> Sent: Saturday, July 02, 2005 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sipura SPA2000 behind NAT
> 
> Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
> adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
> 
> 
> ___ HOME ___   OFFICE 
> SPA2000 <---> Linux Box   <--> Asterisk Box
> 192.168.0.253192.168.0.1 eth1  200.93.xxx.a
>  200.93.xxx.b eth0
> 
> My problem is when I try to call to any trunk or extention I can the
> audio when the destination is ringing, but I can hear the voice of the
> person when it reponds. The person in the other side can hear me, but I
> can not hear anything from him. I can not hear the voice prompts for the
> voicemail (*98) or the operator voice, but can leave voice messages to
> other SIP devices and they can hear my messages.
> 
> This is my sip.conf
> [105]
> username=105
> type=friend
> secret=105
> qualify=no
> port=5060
> nat=yes
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> context=from-internal
> canreinvite=no
> callerid="Guilllermo Salas HOME" <105>
> 
> My ext on line 1 of the Sipura is 105, and is registred with the * box:
> -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600
> 
> asterisk*CLI> sip show peer 105
> asterisk*CLI>
> 
>   * Name   : 105
>   Secret   : 
>   MD5Secret: 
>   Context  : from-internal
>   Language : es
>   FromUser :
>   FromDomain   :
>   Callgroup:  (0)
>   Pickupgroup  :  (0)
>   Mailbox  : [EMAIL PROTECTED]
>   LastMsgsSent : 2
>   Dynamic  : Yes
>   Expire   : 4
>   Expiry   : 900
>   Insecure : No
>   Nat  : Always
>   ACL  : No
>   CanReinvite  : No
>   PromiscRedir : No
>   DTMFmode : rfc2833
>   LastMsg  : 0
>   ToHost   :
>   Addr->IP : 200.93.xxx.xb Port 5060
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Username : 105
>   Codecs   : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263)
>   Codec Order  : (g729|g723|gsm|g726|ulaw|alaw|h261|h263)
>   Status   : UNKNOWN
>   Useragent:
>   Full Contact : sip:[EMAIL PROTECTED]:5060
> 
> And this is the output of sip debug peer 105 when I call to *98 (for
> voice messages):
> 
> asterisk*CLI> sip debug peer 105
> SIP Debugging Enabled for IP: 200.93.xxx.xb:5060
> 
> Sip read:
> NOTIFY sip:sip.mydomain.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
> From: Guillermo Salas M ;tag=4f2df183b116b70c
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 4 NOTIFY
> Max-Forwards: 70
> Event: keep-alive
> User-Agent: Sipura/SPA2000-2.0.2
> Content-Length: 0
> 
> 
> 10 headers, 0 lines
> Transmitting (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
> From: Guillermo Salas M ;tag=4f2df183b116b70c
> To: ;tag=as038653dd
> Call-ID: [EMAIL PROTECTED]
> CSeq: 4 NOTIFY
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Content-Length: 0
> 
> 
>  to 200.93.xxx.xb:50

RE : [Asterisk-Users] Colored asterisk -R?

[f6hqz-m]

Asterisk -gc

Best Regards,
Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stefan
Gofferje
Envoyé : samedi 2 juillet 2005 22:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] Colored asterisk -R?


Hi folks,

when I start asterisk directly, I get a colored CLI. When connect to a 
already running asterisk with asterisk -R, it's never colored, despite 
I'm running both from the same console (tty). Is there a way to force 
asterisk -R into color mode?

Regards,
Stefan

-- 
 (o_   Stefan Gofferje  | Linux Systems Specialist
 //\   Reg'd Linux User #247167 | Network Security Specialist
 V_/_  Heckler & Koch - the original point and click interface

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RE: [Asterisk-Users] call forwarding, most basic case

[Mike Hillerbrand]

Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me

I used and it works well. Rather than segregate calls based on caller ID, it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a "0" before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.

Mike.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case


hello all,

i need some help and after trying the wiki i'm even more confused than i
was.

 i'm trying to set up call forwarding and running into problems...
 i want the most basic call forwarding imaginable.

1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.

as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)

so, i know i need an extension like this:

exten =>154,1, Answer
exten => 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =>153,3, Hangup

but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?

of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.

thanks,
 yair
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[Asterisk-Users] Enum or DUNDi

[Waldo Rubinstein]

I've been reading a bit about Enum and DUNDi and still have something  
not very clear to me.


This is a HYPOTHETICAL scenario:

I have 4 asterisk servers. All of them are handling registrations of  
both SIP and IAX2 UAs. SIP agents are being load balanced by  
something like SER. I have another server in charge of load balancing  
IAX2 UAs registration (some sort of dynamic firewall telling the  
remote host what IP address to connect to) (may be there is already a  
IAX2 load balancer out there).


The idea behind this is that any agent (whether it's SIP or IAX2) can  
register to any of the asterisk servers at any given time, given the  
load on the servers. Therefore, I would assume that all asterisks  
servers have a "replica" of the sip.conf, iax.conf, and some  
intelligent dial plan that would be able to handle all these.


I THINK the questions I have are:
1) Since all asterisk servers can, at any time, accept registration  
from any SIP or IAX2 client, can they all have the same sip.conf and  
iax2.conf?
2) If the answer to question 1 is yes, then if I use something like  
ENUM or DUNDi so that all asterisk servers can "learn" where the  
actual clients are, will the asterisk servers announce their  
knowledge once the client registers on the asterisk server, or will  
the fact that their configuration is "hard-coded" in the sip.conf and  
iax2.conf assumes that all the asterisks servers will simply say, I  
can route call to this "address", even if that client has not  
actually registered on a particular server?


I hope my questions were clear.

Thanks,
Waldo
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[Asterisk-Users] Colored asterisk -R?

[Stefan Gofferje]

Hi folks,

when I start asterisk directly, I get a colored CLI. When connect to a 
already running asterisk with asterisk -R, it's never colored, despite 
I'm running both from the same console (tty). Is there a way to force 
asterisk -R into color mode?


Regards,
Stefan

--
(o_   Stefan Gofferje  | Linux Systems Specialist
//\   Reg'd Linux User #247167 | Network Security Specialist
V_/_  Heckler & Koch - the original point and click interface

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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Ade Agbero]

I used the astcc.agi you posted on your website, and changed the "BRANDS" INC value from 6 to 60.
 
So, I guess all you need to do now is incorporate your astcc.agi file into the CVS version.
 
Thanks goes to you and those who contributed and responded to my HELP HELP HELP plea.
 
From a satisfied ASTCC user,
 
Ade.Darren Wiebe <[EMAIL PROTECTED]> wrote:
Sorry I was not available yesterday. It was Canada Day and we got to celebrate Alberta's centenial. What did you wind up doing to get it working?Darren Wiebe[EMAIL PROTECTED]Ade Agbero wrote:> Problem resolved with Astcc, certified fully working.>> */Juan Luis Moyano <[EMAIL PROTECTED]>/* wrote:>> Ade Agbero wrote:> > Finally, We have lift off, a shaky one though.> >> > I deleted my Astcc.gi and replaced it with Darren's copy posted> on his> > website and I have finally been able to get something recorded> as BILLCOST.> >>> I got it working too here with Darren's astcc.agi. And billing as> expected so finally It's working. It would be nice if someone could> update the cvs with Darren's astcc.agi, because the
  current
 one> doesn't> work, even patched.. it gets worse. Thanks for your attention Darren!>> -- > Juan Luis Moyano> [EMAIL PROTECTED]>> ___> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>> > Yahoo! Messenger > > NEW - crystal clear PC to PC calling worldwide with voicemail > >>>>>___>Asterisk-Users mailing list>Asterisk-Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk-users>To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Telephoning Announcements -- Suggestions?

[Michael Welter]

Scott Nelson wrote:
In the subdivision where I live, we have a well that time to time has  
problems.  


How about just fix the well :-)
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Re: [Asterisk-Users] Telephoning Announcements -- Suggestions?

[Darren Wiebe]

If you do add a bounty, I'll add a little bit $50US or so to it.  I know 
there are others that have written or would like a script like this as 
it was discussed a few months ago on this list I believe.


Darren Wiebe
[EMAIL PROTECTED]

Dean Collins wrote:


Hi Scott, everything thing you are looking to do is possible.

You might need to offer a bounty for someone if you don't feel
comfortable to do it yourself but checkout the call agi scripts.

Cheers,
Dean


 


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Nelson
Sent: Saturday, 2 July 2005 4:05 PM
To: Asterisk Users Mailing List - Asterisk Users
Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions?

In the subdivision where I live, we have a well that time to time has
problems.  Currently, our trustees call me, I take a message, and
then call the people on our "phone calling tree" and give them the
message.  They, in turn, pass the message on to the rest of the
residents.

We have a few problems with the way this works -- people don't bother
to relay the message, or, when they do relay the message, they get
the message wrong.

I was thinking about using an Asterisk server -- kind of a "hot line"
for our subdivision.  A trustee can call in and record an
announcement, and the server would phone each resident and play the
announcement to them.  At other times we can use the server to play
"up and coming events" and so on.

Obviously, I need to create the menus and authenticate the trustees,
record the message, and have that trigger an agi to create outbound
dial files -- and some sort of cron or agi job to get statistics of
who has acknowledged the message and who hasn't so that the trustee
can phone the resident themselves.

I'm thinking about a small PC located at one of the trustees' house
(or perhaps my own) with one phone line interface.  Alas, no high
speed internet here so I can't use VOIP instead of a POTS phone line.

Does anyone have suggestions/tips/cautions for me?  Are there any
applications that might help me?  Should I create a program that
makes the calls via the manager interface, etc.

Thanks,
Scott Nelson
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[Asterisk-Users] Sipura SPA2000 behind NAT

[Guillermo Salas M]

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:


___ HOME ___   OFFICE 
SPA2000 <---> Linux Box   <--> Asterisk Box
192.168.0.253192.168.0.1 eth1  200.93.xxx.a
 200.93.xxx.b eth0

My problem is when I try to call to any trunk or extention I can the
audio when the destination is ringing, but I can hear the voice of the
person when it reponds. The person in the other side can hear me, but I
can not hear anything from him. I can not hear the voice prompts for the
voicemail (*98) or the operator voice, but can leave voice messages to
other SIP devices and they can hear my messages.

This is my sip.conf
[105]
username=105
type=friend
secret=105
qualify=no
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Guilllermo Salas HOME" <105>

My ext on line 1 of the Sipura is 105, and is registred with the * box:
-- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600

asterisk*CLI> sip show peer 105
asterisk*CLI>

  * Name   : 105
  Secret   : 
  MD5Secret: 
  Context  : from-internal
  Language : es
  FromUser :
  FromDomain   :
  Callgroup:  (0)
  Pickupgroup  :  (0)
  Mailbox  : [EMAIL PROTECTED]
  LastMsgsSent : 2
  Dynamic  : Yes
  Expire   : 4
  Expiry   : 900
  Insecure : No
  Nat  : Always
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 200.93.xxx.xb Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Username : 105
  Codecs   : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263)
  Codec Order  : (g729|g723|gsm|g726|ulaw|alaw|h261|h263)
  Status   : UNKNOWN
  Useragent:
  Full Contact : sip:[EMAIL PROTECTED]:5060

And this is the output of sip debug peer 105 when I call to *98 (for
voice messages):

asterisk*CLI> sip debug peer 105
SIP Debugging Enabled for IP: 200.93.xxx.xb:5060

Sip read:
NOTIFY sip:sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 4 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0


10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: ;tag=as038653dd
Call-ID: [EMAIL PROTECTED]
CSeq: 4 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 200.93.xxx.xb:5060
Destroying call '[EMAIL PROTECTED]'

asterisk*CLI>

Sip read:
NOTIFY sip:sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 6 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0


10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: ;tag=as5099fa8f
Call-ID: [EMAIL PROTECTED]
CSeq: 6 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 200.93.xxx.xb:5060
Destroying call '[EMAIL PROTECTED]'
asterisk*CLI>


I dial *98 to get into the voice message system:

asterisk*CLI>

Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-600583f3
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: ;tag=as65eec750
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 70
Contact: Guillermo Salas M 
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0


10 headers, 0 lines
asterisk*CLI>

Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-ec22067b
From: Guillermo Salas M ;tag=4f2df183b116b70c
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="105",realm="asterisk",nonce="47a68adb",uri="sip:[EMAIL 
PROTECTED]",algorithm=MD5,response="8e60f592df094f9b852a59544b9da384"
Contact: Guillermo Salas M 
Expires: 240
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 422
Content-Type: application/sdp

v=0
o=- 12384 12384 IN IP4 192.168.0.253
s=-
c=IN IP4 192.168.0.253
t=0 0
m=audio 16468 RTP/AVP 4 0 2 8 18 96 97 98 100 101
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

13 headers, 19 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (NAT)
Found user '105'
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 18
Found RTP

RE: [Asterisk-Users] Telephoning Announcements -- Suggestions?

[Dean Collins]

Hi Scott, everything thing you are looking to do is possible.

You might need to offer a bounty for someone if you don't feel
comfortable to do it yourself but checkout the call agi scripts.

Cheers,
Dean


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Scott Nelson
> Sent: Saturday, 2 July 2005 4:05 PM
> To: Asterisk Users Mailing List - Asterisk Users
> Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions?
> 
> In the subdivision where I live, we have a well that time to time has
> problems.  Currently, our trustees call me, I take a message, and
> then call the people on our "phone calling tree" and give them the
> message.  They, in turn, pass the message on to the rest of the
> residents.
> 
> We have a few problems with the way this works -- people don't bother
> to relay the message, or, when they do relay the message, they get
> the message wrong.
> 
> I was thinking about using an Asterisk server -- kind of a "hot line"
> for our subdivision.  A trustee can call in and record an
> announcement, and the server would phone each resident and play the
> announcement to them.  At other times we can use the server to play
> "up and coming events" and so on.
> 
> Obviously, I need to create the menus and authenticate the trustees,
> record the message, and have that trigger an agi to create outbound
> dial files -- and some sort of cron or agi job to get statistics of
> who has acknowledged the message and who hasn't so that the trustee
> can phone the resident themselves.
> 
> I'm thinking about a small PC located at one of the trustees' house
> (or perhaps my own) with one phone line interface.  Alas, no high
> speed internet here so I can't use VOIP instead of a POTS phone line.
> 
> Does anyone have suggestions/tips/cautions for me?  Are there any
> applications that might help me?  Should I create a program that
> makes the calls via the manager interface, etc.
> 
> Thanks,
> Scott Nelson
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Telephoning Announcements -- Suggestions?

[Scott Nelson]

In the subdivision where I live, we have a well that time to time has  
problems.  Currently, our trustees call me, I take a message, and  
then call the people on our "phone calling tree" and give them the  
message.  They, in turn, pass the message on to the rest of the  
residents.


We have a few problems with the way this works -- people don't bother  
to relay the message, or, when they do relay the message, they get  
the message wrong.


I was thinking about using an Asterisk server -- kind of a "hot line"  
for our subdivision.  A trustee can call in and record an  
announcement, and the server would phone each resident and play the  
announcement to them.  At other times we can use the server to play  
"up and coming events" and so on.


Obviously, I need to create the menus and authenticate the trustees,  
record the message, and have that trigger an agi to create outbound  
dial files -- and some sort of cron or agi job to get statistics of  
who has acknowledged the message and who hasn't so that the trustee  
can phone the resident themselves.


I'm thinking about a small PC located at one of the trustees' house  
(or perhaps my own) with one phone line interface.  Alas, no high  
speed internet here so I can't use VOIP instead of a POTS phone line.


Does anyone have suggestions/tips/cautions for me?  Are there any  
applications that might help me?  Should I create a program that  
makes the calls via the manager interface, etc.


Thanks,
Scott Nelson
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RE: [Asterisk-Users] play message to callee before connect toincomingcall

[Roland Zagler]

sorry for the misunderstanding, robert!

the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should be
connected
to the sip phone 100.

the behaviour is just like MoH, but the problem is, that the caller has
to hear a
soundfile from the beginning, and MoH does not do this as the sound
files inside
the directory specified in musiconhold.conf are started when asterisk is
started.

an idea i had was to write an agi script which is called in dialplan
before i issue a
playback command to the caller, which is producing a callfile to call
sip phone 100,
but how can i "connect" these 2 calls afterwards? is there a possibility
to connect
the answered sip phone 100 to the incoming call (that is still listening
to the playback
command)?

roland

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Saturday, July 02, 2005 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] play message to callee before connect
toincomingcall


On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:

> Thank you, Robert!
>
> The announcementfile plays well, but at Dial-option "m" i have to
> specify a MoH class,
> that is something i cannot use (as i wrote in my post).

Noted, which is why I offered option two.

>
> Background command waits for a user input, but the caller should be
> connected to
> SIP Phone 100 after it has answered and the announcement has been
> played.

But it doesn't REQUIRE input. Background completes when then sound file 
ends. Are you saying you want to move on to announcing the call to the 
callee as soon as it comes in while the caller is listening to the 
soundfile?

I was following your sequential steps in your post, but if you intend 
to fork the process and be doing two things at once, then it's more 
complex.


> Before
> connecting to SIP Phone 100 the caller should hear a soundfile...
>
> wiki says nothing about an Dial-option to play a soundfile to the 
> caller
> ;-(

Sure it does... BACKGROUND.

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Re: [Asterisk-Users] What to use h323 or oh323 ???

[Tzafrir Cohen]

On Sat, Jul 02, 2005 at 09:13:47AM -0700, Adeel -31 wrote:
> I m new to asterisk n i've got an IP phone that supports h323 protocol 
> but i dont know how to configure asterisk to use it... 

There are currently a number of options for h323 support:

- The original chan_h323 . Works only with HEAD. Requires some specific
  version of openh323 and pwlib

- chan_oh323 . There is a certain version that works with Stable (and it
  should be availble in Debian debs). Another one in HEAD. Requires
  it's own specific version of the monster (openh323 and pwlib, IIRC)

- chan_ooh323c . A new h323 stack. Does not require the monster. However
  it is probably the least tested of all options

- chan_woomra: an attempt to hide the monster. Still requires openh323
  and pwlib and something called opal on top of it, but all asterisk
  sees is a simple wrapper that talkes to that server through a
  pipe/socket. Should work with both HEAD and stable.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] IAX DTMF Problem...

[Julio Arruda]

Denis Galvão - iSolve wrote:

IAX doesn't use INBAND DTMF.

Denis Galvão.


Denis,
A clarification, I hope, just to make Mark aware of the small difference.
IAX sends DTMF in the signaling 'stream', that happens to follow the 
same path as the media.
But, in IAX DTMF is not sent as voice payload (as the typical inband 
when is being used in SIP as one example), just happen to be sent "in 
the same flow" from source host/port to destination host/port.
In SIP, when we use "inband", in fact, the DTMF tone will be treated as 
a voice 'piece', and will be sent in RTP packets without any special 
considerations :-)
From what I understand, that is one of the reasons with SIP inband 
doesn't mix well with any codec other than G.711.

(compression could distort the tone).
Mark, you may want to ask your provider to check the same kind of trace 
you did in your side.
(after all, it seems much like you ARE sending the DTMF to them, is not 
like will get distorted :-)




On 01 de jul de 2005, at 03:23, Mark Edwards wrote:




Hi.

Probably been asked before, but my IAX provider assures me its not  
their problem


I have a IAX connection to a peer providing a DID. I am dialing up  my 
number, seeing the DTMF tones come down the line, and the * IVR  is 
just ignoring them.


IAX debug output is:

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1
   Timestamp: 02608ms  SCall: 00016  DCall: 3  [ 210.80.176.12:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX  
Subclass: ACK

   Timestamp: 02608ms  SCall: 3  DCall: 00016 [210.80.176.12:4569]

for a press of "1"

I am assuming this is the DTMF inband problem, but I appear unable  to 
convince my provider.


Can I work around this on * or do I have to go back to SIP?


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Re: [Asterisk-Users] Linux Firewall Question

[Tzafrir Cohen]

On Fri, Jul 01, 2005 at 12:15:06PM -0400, Michael Stahl wrote:
> You should be able to do a good job with IPTABLES which is included in
> FC3.  You can limit source & destp IP and protocol, etc.
> 
> Type "man iptables | more" for more details...

Which will not get you anywhere. There are a number of relevant HOWTOs
on this subject. One reasonable starting point is iptables (actually:
netfilter)'s homepage:

  http://netfilter.org/documentation/index.html#documentation-howto

Also worth mentioning is the command with the confusing name
'iptables-save', which dumps the current iptables rules and
iptables-estore which restores from those rules. iptables-save is handy
as a simple rules browser.

But you should generate those rules in some automated way, because
you'll need to somehow change them from a remote location when you'll
least expect that. This is why I'd avoid most of the graphical apps such
as firestarter.

Shorewall mentioned earlier is also very handy. One of its design goals
is to save you from mistakes and make it much less probable that you
lock yourself out.

However I chose to avoid it beccause it creates relatively complex and 
"expensive" rules: with Asterisk you'd want ti minimize the amounts of
tests each good voip packet is subject to before it is allowed in.
Shorewall's rules will subject the packet to many unnecessary rules.

It may also be confusing for a simple one-interface server setup. That's
why I have decided to write one myself.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] play message to callee before connect toincoming call

[Robert Goodyear]

On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:


Thank you, Robert!

The announcementfile plays well, but at Dial-option "m" i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).


Noted, which is why I offered option two.



Background command waits for a user input, but the caller should be
connected to
SIP Phone 100 after it has answered and the announcement has been
played.


But it doesn't REQUIRE input. Background completes when then sound file 
ends. Are you saying you want to move on to announcing the call to the 
callee as soon as it comes in while the caller is listening to the 
soundfile?


I was following your sequential steps in your post, but if you intend 
to fork the process and be doing two things at once, then it's more 
complex.




Before
connecting to SIP Phone 100 the caller should hear a soundfile...

wiki says nothing about an Dial-option to play a soundfile to the 
caller

;-(


Sure it does... BACKGROUND.

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Re: [Asterisk-Users] Resolving groupcalls

[Brian West]

You can also set anything you wish into the CDR variables.  We came  
up with the whole CDR variable thing for this exact purpose.  Check  
cdr_custom to log it like you want.


ie Set(CDR(GROUP)=${GROUPCALL})



/b
PS don't for get to come to cluecon!

On Jun 30, 2005, at 4:15 AM, Chris Coulthurst wrote:


Oops, sent that last one prematurely!

How about the accountcode setting?  You could get user information  
from

that, right?

Maybe you could send:

Asterisk -rx 'show channels'

..and when you get the data, you'd know which channels are up and  
alive

(full names).

You could then re-run the command with the channel information:

Asterisk -rx 'show channel SIP/201-ec69'

..you'd get a dump, with the end looking something like this:
  CDR Variables:
level 1: clid="Chris Office" <201>
level 1: src=201
level 1: dst=18009427433
level 1: dcontext=unlimited
level 1: channel=SIP/201-ec69
level 1: dstchannel=IAX2/provider-7
level 1: lastapp=Dial
level 1: lastdata=iax2/[EMAIL PROTECTED]/2047622726
level 1: start=2005-06-30 02:10:35
level 1: answer=2005-06-30 02:10:38
level 1: end=2005-06-30 02:10:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: accountcode=019284718233 <<--account code
unique to the user
level 1: uniqueid=1120122635.400


Anyway, maybe something like that...

Chris Coulthurst
[EMAIL PROTECTED]



|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm trying to write a tool, which shows me the state of the current
|calls. For this purpose I'm reading from Pipe the Asterisk output and
|parse it... asterisk -vr | mytool
|
|However, the problem ist how to get the information about who got  
this

|call in the group. The Zap channels are assigned dynamical.
|Only thing I
|can see which channel is connect to the caller but not who is
|using the
|channel.
|
|I know there is the CDR output in Master.csv. But it shows me
|the same.
|The other problem with CDR is, that it shows me the Info only
|after the
|call is finished. That's why I'm trying to parse the asterisk output.
|
|My extensions.conf looks like this..
|GROUPCALL => Zap/g2/1200021&Zap/g2/1200022&Zap/g2/1200023
|.
|.
|exten => s,1,Dial(${GROUPCALL})
|
|
|___
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|UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] play message to callee before connect toincoming call

[Roland Zagler]

Thank you, Robert!

The announcementfile plays well, but at Dial-option "m" i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).

Background command waits for a user input, but the caller should be
connected to
SIP Phone 100 after it has answered and the announcement has been
played. Before
connecting to SIP Phone 100 the caller should hear a soundfile...

wiki says nothing about an Dial-option to play a soundfile to the caller
;-(


Roland Zagler 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Saturday, July 02, 2005 8:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] play message to callee before connect
toincoming call

On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote:

> Hello,
>
> i try to do the following:
>
> 1) call comes in on extension 999
> 2) caller should hear music (NOT MoH!!!)
> 3) a call should be initiated to SIP Phone 100
> 4) when SIP Phone 100 is answered, a sound file should be played to
the
> user at SIP Phone 100
> 5) the incoming call (at extension 999) should be connected to SIP 
> Phone
> 100
>

Rough pseudo-code follows, experiment and report your results to the 
list:

999 dial(SIP/100|20|m(soundfile)A(announcementfile))

OR

999 background(soundfile)
999 dial(SIP/100|20|A(announcementfile))



-- 
Robert Goodyear
Brand Up LLC
http://www.brand-up.com



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Re: [Asterisk-Users] play message to callee before connect to incomingcall

[Robert Goodyear]

On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote:


try this one

exten => 999,1,Answer()
exten => 999,2,playback(~.mp3)
exten => 999,3,dial (sip/100)
exten => 999,4,playbackground(~.mp3)
exten => 999,h,Hangup()


not sure abt playbackground should be before the dial command or after




Mahmoud: you don't pass file extension to the playback app, and there's 
no such app called playbackground. Plus the OP wanted the announcement 
to hit the callee solely.


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[Asterisk-Users] call forwarding, most basic case

[Yair Hakak]

hello all,

i need some help and after trying the wiki i'm even more confused than i was.

 i'm trying to set up call forwarding and running into problems...
 i want the most basic call forwarding imaginable.

1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.

as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)

so, i know i need an extension like this:

exten =>154,1, Answer 
exten => 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM}) 
exten =>153,3, Hangup

but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?

of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.

thanks,
 yair
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RE: [Asterisk-Users] play message to callee before connect to incomingcall

[Mahmoud Badran]

try this one
 
exten => 999,1,Answer()
exten => 999,2,playback(~.mp3)
exten => 999,3,dial (sip/100)
exten => 999,4,playbackground(~.mp3)
exten => 999,h,Hangup()
 
 
not sure abt playbackground should be before the dial command or after

 


From: [EMAIL PROTECTED] on behalf of Roland Zagler
Sent: Sat 7/2/2005 8:23 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] play message to callee before connect to incomingcall



Hello,

i try to do the following:

1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP Phone 100
5) the incoming call (at extension 999) should be connected to SIP Phone
100

any suggestions on how to implement this in an easy way?

Thanks in advance,
Roland Zagler
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Re: [Asterisk-Users] play message to callee before connect to incoming call

[Robert Goodyear]

On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote:


Hello,

i try to do the following:

1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP Phone 100
5) the incoming call (at extension 999) should be connected to SIP 
Phone

100



Rough pseudo-code follows, experiment and report your results to the 
list:


999 dial(SIP/100|20|m(soundfile)A(announcementfile))

OR

999 background(soundfile)
999 dial(SIP/100|20|A(announcementfile))



--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com



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Re: [Asterisk-Users] Is it possible to setup group voicemail in Asterisk?

[Andrew Latham]

exten => 1234,5,Voicemail(u,1234&2345&3456&4567)

As you can see the same voicemail will go to all the users.

On 7/2/05, Leo Burd <[EMAIL PROTECTED]> wrote:
> Hello there,
> 
> I'm a new Asterisk user and I wonder if it is possible to associate a
> voicemail box with a group of users, i.e., a single recorded message is
> sent to everyone in that group.  If so, where can I find more
> information about that?
> 
> Thanks in advance,
> 
> Leo Burd
> 
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 

Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!

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Re: [Asterisk-Users] What to use h323 or oh323 ???

[Bashir Ullah - www.Lamsre.Com]

Hi Adeel
 
http://www.inaccessnetworks.com/projects/asterisk-oh323
 
Please visit there, you will find 
your way.
 
Bashir

  - Original Message - 
  From: 
  Adeel -31 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Saturday, July 02, 2005 9:13 
  AM
  Subject: [Asterisk-Users] What to use 
  h323 or oh323 ???
  
  I m new to asterisk n i've got an IP phone that supports h323 
  protocol but i dont know how to configure asterisk to use it... i m 
  comfortable in using sip & iax softphones but there is no 
  h323.conf in /etc/asterisk/    i read that i've to compile some 
  files but i m confused regarding h323 & oh323  .. which one 
  should i use.. plz tell me or atleast give some helpful link
  __Do You 
  Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around 
  http://mail.yahoo.com 
  
  

  ___Asterisk-Users 
  mailing 
  listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
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[Asterisk-Users] play message to callee before connect to incoming call

[Roland Zagler]

Hello,

i try to do the following:

1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP Phone 100
5) the incoming call (at extension 999) should be connected to SIP Phone
100

any suggestions on how to implement this in an easy way?

Thanks in advance,
Roland Zagler
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RE: [Asterisk-Users] Is it possible to setup group voicemail inAsterisk?

[Roland Zagler]

Hi Leo,

here's a suggestion:

in your dialplan (extensions.conf) send multiple users to the same
mailbox (e.g. 999) if they do not pick up within 30 seconds:

; SIP Phone 100, Tom
exten => 100,1,Dial(SIP/100,30)
exten => 100,2,VoiceMail(999)

; SIP Phone 200, Eric
exten => 200,1,Dial(SIP/200,30)
exten => 200,2,VoiceMail(999)

and in your voicemail.conf you do not specify a single user's email
address
([EMAIL PROTECTED] or [EMAIL PROTECTED]), but a email group created in your
email system
then the voicemail file is sent to all the email-addresses in this
group.

this is not an asterisk issue, you should take a look at the manual of
your
email system on how to create groups.

regards,
roland

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Burd
Sent: Saturday, July 02, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Is it possible to setup group voicemail
inAsterisk?

Hello there,

I'm a new Asterisk user and I wonder if it is possible to associate a 
voicemail box with a group of users, i.e., a single recorded message is 
sent to everyone in that group.  If so, where can I find more 
information about that?

Thanks in advance,

Leo Burd

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[Asterisk-Users] Is it possible to setup group voicemail in Asterisk?

[Leo Burd]

Hello there,

I'm a new Asterisk user and I wonder if it is possible to associate a 
voicemail box with a group of users, i.e., a single recorded message is 
sent to everyone in that group.  If so, where can I find more 
information about that?


Thanks in advance,

Leo Burd

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[Asterisk-Users] (Simple?) ENUM Question

[Eric Wieling aka ManxPower]

I've been doing some reading on ENUM and am almost ready to start 
testing with it.  However, I have a question.


As I understand things the following ENUM entry would return info for 
all telephone numbers of any length beginning with 00393.  The Asterisk 
pattern would be _00393.  (notice the dot)


*.3.9.3.0.0 IN  NAPTR 100 100 "u" "SIP+E2U" "!^+*00393(.*)!sip:[EMAIL 
PROTECTED]" .

I would like to know if it's possible to have an ENUM entry that would 
be the equiv of the Asterisk pattern _21XX  (notice no dot).


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Re: [Asterisk-Users] Make Webvmail Error

[Andrew C. Brown]

Manjit Riat wrote:
> 
> 
> I did a make webvmail and I get the following error on redhat 9.0
> 
>  
> 
> No HTTP directory
> 
> make : *** [webvmail] Error 1
> 
>  
> 
> I have the perl-suidperl rpm installed and apache installed
> 
>  
> 
> Thanx .

The webvmail make script isn't terribly intelligent about finding your
web directory, which varies alot from one linux distrib to the next. I'm
looking into making a patch for it to parameterize a little better. If
you go into the make ("Makefile") and search for "$HTTPDIR" you can sort
of see where it is expecting to install it and change it by hand for now
if you want.
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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Darren Wiebe]

Sorry I was not available yesterday.  It was Canada Day and we got to 
celebrate Alberta's centenial.  What did you wind up doing to get it 
working?


Darren Wiebe
[EMAIL PROTECTED]

Ade Agbero wrote:


Problem resolved with Astcc, certified fully working.

*/Juan Luis Moyano <[EMAIL PROTECTED]>/* wrote:

Ade Agbero wrote:
> Finally, We have lift off, a shaky one though.
>
> I deleted my Astcc.gi and replaced it with Darren's copy posted
on his
> website and I have finally been able to get something recorded
as BILLCOST.
>

I got it working too here with Darren's astcc.agi. And billing as
expected so finally It's working. It would be nice if someone could
update the cvs with Darren's astcc.agi, because the current one
doesn't
work, even patched.. it gets worse. Thanks for your attention Darren!

-- 
Juan Luis Moyano

[EMAIL PROTECTED]

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Yahoo! Messenger 
 
NEW - crystal clear PC to PC calling worldwide with voicemail 
 





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Re: [Asterisk-Users] IAX DTMF Problem...

[Denis Galvão - iSolve]

IAX doesn't use INBAND DTMF.

Denis Galvão.


On 01 de jul de 2005, at 03:23, Mark Edwards wrote:




Hi.

Probably been asked before, but my IAX provider assures me its not  
their problem


I have a IAX connection to a peer providing a DID. I am dialing up  
my number, seeing the DTMF tones come down the line, and the * IVR  
is just ignoring them.


IAX debug output is:

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF 
Subclass: 1
   Timestamp: 02608ms  SCall: 00016  DCall: 3  
[ 210.80.176.12:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX  
Subclass: ACK

   Timestamp: 02608ms  SCall: 3  DCall: 00016 [210.80.176.12:4569]

for a press of "1"

I am assuming this is the DTMF inband problem, but I appear unable  
to convince my provider.


Can I work around this on * or do I have to go back to SIP?

Mark




--
regards,

Mark P. Edwards
TEL:+61 408 601 107
SKYPE: mark.p.edwards
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[Asterisk-Users] What to use h323 or oh323 ???

[Adeel -31]

I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones but there is no h323.conf in /etc/asterisk/    i read that i've to compile some files but i m confused regarding h323 & oh323  .. which one should i use.. plz tell me or atleast give some helpful link__Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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RE: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing

[Robert Webb]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tom Rymes
> Sent: Friday, July 01, 2005 11:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom
incoming
> routing
>
> A few things to followup on my earlier post:
>
> 1.) Definitely put the [tdm-in] context in the file "/etc/asterisk/
> extensions_custom.conf". That way your changes will not get
overwritten.
> 2.) I am still unable to make call waiting on the incoming ZAP line
> work, b/c I have not thought up a good way to make this happen.
> hasn't anyone done this before?
> 3.) When setting up the ZAP trunk, I found it usefull to put "w" in
> the dial prefix field to force the system to wait for the dial tone.
> If I didn't do that, I could not call out on the ZAP Channel.
>
> Tom
>

Tom,

  Actually, it is the extensions_custom.conf that DOES get overwritten.
Unles things have completely changed since version 1.0. You should be
using the extensions.conf to place all your permamnant changes. The
_custom was there to tell you that those are the custom settings for
[EMAIL PROTECTED]



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RE: [Asterisk-Users] passing through MWI info from SBC

[Robert Webb]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jon Radon
> Sent: Saturday, July 02, 2005 10:49 AM
> To: andrew matthews; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] passing through MWI info from SBC
>
> Woah woah woah.. why not just disable SBC voicemail and have asterisk
> handle it?  I don't understand why you would go to such great lengths
> when you can just have Asterisk deal with it.
>

Because in most cases, the POTS provider will not disable voicemail on a
per number basis as it is a part of the 'package'. So there really isn't
an option to do that.

The other issue is that with call waiting, if you do not answer the call
there is no way to have Asterisk handle the voicemail. So here is where
you would still need the pass through in order for the subscriber to
know there was a message.

For me, I just got a $5 per month DID and forwarded all my POTS call to
it. I get up to two simultaneous incoming calls that Asterisk handles
completely. It also includes a voicemail system where if my connection
goes down or I exceed those two calls, then if someone leave a message,
it gets emailed to me. The other benefit is that I still have my POTS
line and use it for all my local outgoing calls and use a 1.3 cent per
minute provider for all LD calls and I still save money by not spending
that extra $20 a month for unlimited LD on my POTS line.

Has worked great so far. Only down side is if my DID connection drops, I
have no way to call into the house. But, I just use the cell to call the
wife in that case.



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Re: [Asterisk-Users] E3 card

[Andrew Latham]

Just get one of these.

"The PCI 921-CDS is a low-cost channelized DS3 WAN adapter that can be
used in ImageStream's Industrial Series routers or OEM products
running Linux. The PCI 921-CDS can individually address all of the
DS0s and T1s in a DS3, and it can be used in a wide range of
applications including WAN and Internet routing, Voice-over-IP (VoIP),
medical imaging, and digital video on demand."

http://imagestream.com/PCI_921-CDS.html




On 7/2/05, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Eric Wieling aka ManxPower wrote:
> 
> > Kevin P. Fleming wrote:
> >
> >> Tamas J wrote:
> >>
> >>> Does anybody know what will be the around price for the announced E3
> >>> card from Digium? When is it planned to be ready?
> >>
> >>
> >>
> >> Pricing and release date have not been announced at this time.
> >
> >
> > The product has also been removed from Digium's web site, as far as I
> > can tell.
> >
> Perhaps they decided the market is not that big, and reduced the
> priority of the card. If so, they are not the only ones to do this.
> 
> Regards,
> Steve
> 
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-- 

Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!

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RE: [Asterisk-Users] asterisk showing more than once on ps

[Mr. James W. Laferriere]

Hello All ,

On Sat, 2 Jul 2005, Michael Stahl wrote:

The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk

In safe_asterisk, the program is started with -c by default (console on
TTY9).

That explains why it is starting with a console, but not why it's
running so many times!  Here is what my system (FC3) shows:

[EMAIL PROTECTED] sbin]# ps ax | grep asterisk
3371 ?S  0:00 /bin/sh /usr/sbin/safe_asterisk
3417 ?S  0:00 asterisk -vvvg -c
6846 ?S  0:00 asterisk -vvvg -c
6848 ?S  0:00 asterisk -vvvg -c
6849 ?S  0:00 asterisk -vvvg -c
6850 ?S  0:00 asterisk -vvvg -c
6853 ?S  0:01 asterisk -vvvg -c
6854 ?S  0:00 asterisk -vvvg -c
8479 pts/1S+ 0:00 grep asterisk


Can anyone explain why asterisk is being launched 7 times?

Thanks,
OCG

-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk showing more than once on ps


Do not know why, but have noticed redhat = 1, and debian = many

Not quite. RedHat Enterprise also = many at times, depending on number
of concurrent calls; usually one when idle. Maybe it has something to do
with kernel 2.4 vs 2.6 and how threads show up in ps.

--Luki


Below ps is from a * server on slackware 10.0 using the
command to start(**) .  So I am not sure the '-c' is what is
creating the multiple threads .  linux-2.6 issue maybe ?
Hth ,  JimL

# ps -auxww | grep aster
root   115  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   123  0.0  1.1 11916 5944 ?SJun30   0:01 
/usr/sbin/asterisk -d -v -v -v
root   125  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   130  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   131  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   132  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   139  0.2  1.1 11916 5944 ?SJun30   6:08 
/usr/sbin/asterisk -d -v -v -v
root   155  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   156  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   157  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v
root   158  0.0  1.1 11916 5944 ?SJun30   0:00 
/usr/sbin/asterisk -d -v -v -v

(**)
# after expansion of variables . 
/usr/sbin/asterisk -d -v -v -v >> /var/log/asterisk/debug


# sudo asterisk -V
Asterisk CVS-HEAD-05/01/05-14:10:09

# uname -a
Linux asterisk-1 2.6.11.8 #1 Sun May 1 12:04:14 MDT 2005 i686 unknown unknown 
GNU/Linu

--
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
+--+
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Re: [Asterisk-Users] passing through MWI info from SBC

[Jon Radon]

Woah woah woah.. why not just disable SBC voicemail and have asterisk
handle it?  I don't understand why you would go to such great lengths
when you can just have Asterisk deal with it.

On 7/1/05, andrew matthews <[EMAIL PROTECTED]> wrote:
> Is there alternative access to voicemail? Like web access?
> 
> If there was then you can write a program to check the site every 5 or 10
> minutes maybe every 30, and parse out the message listing and then send MWI
> to the phones if there is a VM waiting.
> 
> Or
> 
> You could get one of those cheep 20 dollar devices and hook it up to the
> serial port on the asterisk box and then you can use sty to monitor the port
> for changes. Then send the message that way. I'd have to research it a
> little more but it very possible.
> 
> Just a thought :)
> 
> 
> On 7/1/05, Chris Gamble <[EMAIL PROTECTED]> wrote:
> > Does this just sound worse than it is?
> > 
> > >> "With SBC you are out of luck, since Asterisk doesn't detect dialtone 
> (
> > >> it dials blind, sometimes too quickly for the CO to catch the first
> > >> digit, resulting in wrong numbers )) or stutter dialtone either, and 
> > >> reportedly has had any indication of the DC status of a POTS line
> > >> removed due to problems."
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf
> Of John Novack
> > Sent: Friday, July 01, 2005 10:42 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] passing through MWI info from SBC
> > 
> > 
> > Mike Myers wrote:
> > 
> > >Hi..  I am about to replace my aging Nortel Venture
> > >system with an Asterisk system and 6 Polycom IP 501
> > >phones, and a couple sipura 841's for less used areas.
> > >
> > >We have 3 phone lines here.  One is SBC, one Vonage,
> > >and one Voipjet...  One hangup is that I can't figure 
> > >out how to pass through a voicemail waiting indication
> > >from SBC.  This is important because my wife and her
> > >family all exchange voicemails with each other on the
> > >SBC voicemail system.  They can leave messages for 
> > >each other without having the phones ring, etc...  We
> > >have a 2 yr old at home, and her sister has some small
> > >kids too, so that's how they manage to send voicemails
> > >when they are unsure if the kids are sleeping, etc... 
> > >Anyway, preserving this capability of using the SBC VM
> > >and being notified when a message is waiting is
> > >critical for good WAF.
> > >
> > >The vonage line and voipjet line can be intergrated
> > >into the Asterisk VM.  My Nortel venture phones light 
> > >the MWI if any line has VM on it, and the display
> > >tells you which lines have VM waiting.  I would love
> > >to be able to duplicate this function on the Polycom's
> > >and hopefully the Sipura's as well. 
> > >
> > >I've looked for answers on this, but haven't found
> > >one, hence the post.  My apologies if I have missed
> > >something.
> > >
> > >Thanks much,
> > >Mike
> > >
> > >
> > You haven't missed much. 
> > With SBC you are out of luck, since Asterisk doesn't detect dialtone  (
> > it dials blind, sometimes too quickly for the CO to catch the first
> > digit, resulting in wrong numbers )) or stutter dialtone either, and
> > reportedly has had any indication of the DC status of a POTS line
> > removed due to problems.
> > 
> > Only choice would to port the number to a VOIP provider and provide the
> > VM in Asterisk.
> > Similar problem with Vonage VM. 
> > 
> > John Novack
> > 
> > ___
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> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> >   
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> >
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> >   
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> > 
> 
> 
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> 


-- 
Is it something someone said, was it something someone said?
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RE: [Asterisk-Users] asterisk showing more than once on ps

[Michael Stahl]

The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk

In safe_asterisk, the program is started with -c by default (console on
TTY9).

That explains why it is starting with a console, but not why it's
running so many times!  Here is what my system (FC3) shows:

[EMAIL PROTECTED] sbin]# ps ax | grep asterisk
 3371 ?S  0:00 /bin/sh /usr/sbin/safe_asterisk
 3417 ?S  0:00 asterisk -vvvg -c
 6846 ?S  0:00 asterisk -vvvg -c
 6848 ?S  0:00 asterisk -vvvg -c
 6849 ?S  0:00 asterisk -vvvg -c
 6850 ?S  0:00 asterisk -vvvg -c
 6853 ?S  0:01 asterisk -vvvg -c
 6854 ?S  0:00 asterisk -vvvg -c
 8479 pts/1S+ 0:00 grep asterisk


Can anyone explain why asterisk is being launched 7 times?

Thanks,
OCG

-Original Message-
From: Luki [mailto:[EMAIL PROTECTED] 
Sent: Friday, July 01, 2005 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk showing more than once on ps

> Do not know why, but have noticed redhat = 1, and debian = many
Not quite. RedHat Enterprise also = many at times, depending on number
of concurrent calls; usually one when idle. Maybe it has something to do
with kernel 2.4 vs 2.6 and how threads show up in ps.

--Luki

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Re: [Asterisk-Users] Provider Survey

[Chris Mason (Lists)]

List Receiver wrote:

Having used Broadvoice for a while with marginal service, I want to 
move on to another provider. So my question to the List is who is 
good? I know now one service is perfect but somebody out there has to 
be decent. Who have you guys had the best luck with?  

I suggest, if your installation is mission critial at all, you use a 
dialplan setup that has failover between two providers with failover to 
PSTN, and that you consider Teliax and NuFone as Providers. I have found 
these are serious players, they are not lemonade stands, and they offer 
failover to pstn for incoming calls also. I have never had an instance 
where both are down, but my experiene is relatively short.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Error with app_addon_sql_mysql.c

[Sahil Gupta]

Hi People!
Having interesting issues with app_addon_sql_mysql.c:

[EMAIL PROTECTED]:/usr/src/asterisk-addons# make
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include 
-I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 
arguments, but only 3 given

app_addon_sql_mysql.c: In function `del_identifier':
app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use 
in this function)
app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported 
only once

app_addon_sql_mysql.c:164: error: for each function it appears in.)
make: *** [app_addon_sql_mysql.o] Error 1

This is a installation of Slackware 10.1 with Mysql 4.1.12 (source).

Any ideas?

Regards,


Sahil Gupta
VoiceValley

On Fri, 1 Jul 2005, Brian West wrote:

You could have just done "ln -s asterisk-1.0.9 asterisk" and it would have 
fixed that.  It should by default do -I../asterisk


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote:


Marcel van Kaam, Fonetica wrote:



I had the same problem with installing addons. I checked out in the file
cdr_addons_mysql.c what the location of the asterisk.h must be and changed
the cdr_addons_mysql.c to the location of the asterisk.h file.

After this it worked. Also to be sure do: locate asterisk.h to check or 
you

have the file on your system.

Marcel


Yes, that worked. For the record, it had to be

#include "../asterisk-1.0.9/asterisk.h"

--
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(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670
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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC

[Ade Agbero]

Problem resolved with Astcc, certified fully working.Juan Luis Moyano <[EMAIL PROTECTED]> wrote:
Ade Agbero wrote:> Finally, We have lift off, a shaky one though.> > I deleted my Astcc.gi and replaced it with Darren's copy posted on his > website and I have finally been able to get something recorded as BILLCOST.> I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren!-- Juan Luis Moyano[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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 NEW - crystal clear PC to PC
calling worldwide with voicemail
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[Asterisk-Users] Audio delay w/ call forwarding

[Mike Hillerbrand]

I have experienced a * problem with all "forwarded" calls where the inbound
caller cannot hear any audio for 2-4 seconds after the forwarded call is
answered, causing the caller--who cannot hear anything--to think there is no
connection and thus hangs up. If the caller waits a couple of seconds, audio
is restored and everything is OK. The problem didn't seem to be there when I
first set up the server about 2 months ago. After eliminating potential
hardware problems, etc., I figured out that when I change jitterbuffer=yes
to jitterbuffer=no in iax.conf the audio delay problem is solved, but the
call quality degrades. My prior settings were:

[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
mailboxdetail=yes
register => [EMAIL PROTECTED]
dropcount=3
jitterbuffer=yes
maxjitterbuffer=500
maxexcessbuffer=300
canreinvite=no

Any ideas on tweaking iax.conf to optimize call quality, but avoid the audio
delay with forwarded calls?


Regards,

Mike

Michael Hillerbrand

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Re: [Asterisk-Users] Problems loading asterisk .

[Matt Riddell]

Bharat M. Sarvan wrote:

Hello everybody,

I have made a application of my own. (I.e. Def ( )). I am able to
compile the application successfully. And the .so file is created as
well. But when I load asterisk I get the following error.




[Def.so]Jul  1 19:20:06 WARNING[15664]: loader.c:295
ast_load_resource: No key routine in module
/usr/lib/asterisk/modules/Def.so


You need to return the Asterisk GPL Key, have a look at app_skel.c (if I
remember correctly).

I.E.:

char *key()
{
return ASTERISK_GPL_KEY;
}

And from module.h:

#define ASTERISK_GPL_KEY \
"This paragraph is Copyright (C) 2000, Linux Support Services,
Inc. \
In order for your module to load, it must return this key via a function \
called \"key\". Any code which includes this paragraph must be licensed
under \
the GNU General Public License version 2 or later (at your option).
Linux \
Support Services, Inc. reserves the right to allow other parties to
license \
this paragraph under other terms as well."

(Sorry about the line wrapping)

--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Epia C3 Linux

[Tzafrir Cohen]

On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote:
> Anyone know a good distro for an Epia Mobo with the C3 chip?   
> 

Debian, as for any hardware :-p

>  
> 
> I have been trying to get Debian and Gentoo installed (new to me) and so
> far having little luck.  
> 

Debian i386 packages should work on VIA Epia-s as-is. Specifically, see
http://bugs.debian.org/294884

> 
> Does anyone know a good install for this processor/mobo combo?
> 

I have has some success stories of Rapid with it.

-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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[Asterisk-Users] Snom -> Asterisk -> Vegastream

[Neil Bullock]

Looking for some advice from vegastream users if possible?

I am having a nightmare trying to find the best settings for G729 and
G723.1 codecs. My users are using Snom phones. Any recommendations as to
the best codec settings would be very appreciated as trial and error is
proving long and unproductive!

Thanks,

Neil


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Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6

[Andrew Kohlsmith]

On Saturday 02 July 2005 07:57, Rich Adamson wrote:
> Think he's trying to set it on a TDM-fxs module (not fxo). Or did I miss
> something.

Nope I am probably the one who is missing it, it's early here.  :-)

-A.
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Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6

[Rich Adamson]

> On Saturday 02 July 2005 02:04, Keith Caldwell wrote:
> > I just got it working with a 2.6 kernel without any problems,
> > although I'm having a little bit of trouble setting the callerid on it
> 
> You can't set outgoing callerID on analog PSTN lines.  This is  a telephone 
> network limitation, not a Digium hardware limitation.

Think he's trying to set it on a TDM-fxs module (not fxo). Or did I miss
something.


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Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6

[Andrew Kohlsmith]

On Saturday 02 July 2005 02:04, Keith Caldwell wrote:
> I just got it working with a 2.6 kernel without any problems,
> although I'm having a little bit of trouble setting the callerid on it

You can't set outgoing callerID on analog PSTN lines.  This is  a telephone 
network limitation, not a Digium hardware limitation.

-A.
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[Asterisk-Users] PortaOne's Radius client for Asterisk

[Kamran Ahmad]

hello

i m trying to use radius with asterisk
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth

how to fix this patch

8. Make sure that your Asterisk includes all related
bug fixes and patches, namely: 
- SIPGetHeaders for chan_sip (derived from chan_sip2 )


i m using asterisk-1.0.9

extensions.conf
exten =>
_X.,1,SIPGetHeader(SIP_Authorization=Proxy-Authorization)




 
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[Asterisk-Users] editing time to say astcc-noanswer

[wassim darwish]

i dont know how to edit the time "3ms" for ringing
in astcc when it says "there is no body to answer".i
want to change this time to 4ms but i dont know
how.please help please. 



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Re: [Asterisk-Users] Errors Question

[Rich Adamson]

>   Today we have been having some problems with the dchannel of out T1's. 
> I was wondering if there is a way for asterisk to send out an email or 
> page whenan error occurs. Not I know errors happen quite offen for many 
> reasons, but I would like an email sent when there is a TI problem, or 
> specefic other problems. I am using the safe_asterisk to run asterisk 
> and it sends an email when asterisk crashes, which is not very often.

Take a look at using a combination of logging and swatch.
Goggle for 'swatch syslog'.


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Re: [Asterisk-Users] make error for zaptel

[Bob Goddard]

On Friday 01 Jul 2005 16:43, Zoltan Szecsei wrote:
> Bob Goddard wrote:
> >On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
> >>Hi Bob,
> >>Thanks - I'll run with the README idea of yours.
> >>Your comment regarding re-boot however is not valid. I also thought that
> >>was the case and (as I said on the first line of my message) I
> >>specifically rebooted the box. Have to confess I am really flumuxed why
> >>the symbolinc link differs from the uname -r name.
> >
> >I cannot see what the problem is with the output of 'uname -r'!
>
> I'm saying that I though that if uname -r returns:   2.6.11.4-20a-smp
> then I would expect that /usr/src/linux would link to
> linux-2.6.11.4-20a-smp and it does not, it links to linux-2.6.11.4-21.7
>
> see:
>
> gl0:/usr/src # ls -la
[...]

First, learn to trim your posts. I have already read the signature
and I don't need to read it any more.

You will have updated the kernel via Yast. Yast will remove all
previous versions. You have 2 kernels at least partially installed.
Your system is broken. When you updated the kernel, the linux link
was recreated, but Yast did not manage to remove the previous version.
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[Asterisk-Users] Operators Panel for Asterisk

[Thorben Jensen]

IPSwitchBoard Version 0.121 - 02 July 2005

* Extensions can be added to speed dial number. This can be used to dial
speed dial numbers from any phone connected to your asterisk system. This
requires that you configure your dial plan to take advantage of this
feature. See sample Dial Plan in the IPS Manual.

Download: http://ipswitchboard.thorben.dk

_
IPSwitchBoard is a FREE Windows.Net application that will allow you to: 

Unattended/attended transfers. 
Park calls and retrieve/forward them again. 
Organize all your SIP, IAX and Zap extensions (automatically retrieved from
Asterisk). 
Hotel/Call shop Billing module 
Monitor all extensions, queues, agents and Parked Calls. 
Dynamically log extensions in and out of queues. 
Integration with CRM software on the web. 
Browse Call Records and make Charts. 
Record calls and transfer wav files to the PC automatically. 
Set Do Not Disturb on Extensions and give a reason. 
Speed Dialing. Share speed dial number between users of IPS 


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Re: [Asterisk-Users] MOH - request to schdule in the past

[Rich Adamson]

> I have googled this to death, and all I get are reference to the MoH
> needing a Zaptel timing source, and then people saying no they don't
> any more.
> 
>-- Set Response Timeout to 2
> -- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack
> -- Playing 'my-greeting' (language 'en')
>   == CDR updated on SIP/211-57ba
> -- Executing MusicOnHold("SIP/211-57ba", "") in new stack
> -- Started music on hold, class 'default', on SIP/211-57ba
> -- Stopped music on hold on SIP/211-57ba
>   == Spawn extension (webtree, 7, 1) exited non-zero on 'SIP/211-57ba'
> Jul  1 20:19:32 NOTICE[1376]: res_musiconhold.c:309 monmp3thread:
> Request to schedule in the past?!?!
> 
> It seems to work for a little while after a reboot then dies a death. 
> The extension just gets dead air, and the error message is on the CLI,
> it always comes up after a call, and seemingly randomly inbetween.
> 
> Any suggestions where to look?
> 
> extensions.conf has
> exten => _[123],3,Dial(SIP/200&SIP/202&SIP/210&SIP/211&SIP/212|40|m)
> and
> exten => 7,1,musiconhold()
> 
> both do the same thing.
> 
> musiconhold.conf has
> [classes]
> default => mp3:/var/lib/asterisk/mohmp3,-z
> ;loud => mp3:/var/lib/asterisk/mohmp3
> ;random => quietmp3:/var/lib/asterisk/mohmp3,-z

>From your system command line (not asterisk), type 'mpg123' and tell
us what version of mpg123 you're running.

If its not v0.59r or v0.59q, then get one of those installed.
(Lots of notes say v0.59r only, however I've been using v0.59q
on RHv9 and Fedora 3 boxes with no problems.)


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Re: [Asterisk-Users] pattern matching based on callerid, not working

[tim panton]

On 2 Jul 2005, at 08:48, Dinesh Nair wrote:




On 07/02/05 02:15 Matthew Boehm said the following:


according to the wiki, I should be able to do this:
exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,n,Set(CALLERID(number)=281444)
exten => _9./3005,n,Set(CALLERID(number)=281445)
exten => _9./3006,n,Set(CALLERID(number)=281446)



i believe these four should be on the same priority. asterisk  
dialplans will bork if the next priority doesnt match.




Or better yet, fold them into a single line - like:

exten => _9./300[3456],1,Set(CallerID(number)=28144${CALLERIDNUM}:1)

Tim.
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Re: [Asterisk-Users] pattern matching based on callerid, not working

[Dinesh Nair]

On 07/02/05 02:15 Matthew Boehm said the following:

according to the wiki, I should be able to do this:

exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,n,Set(CALLERID(number)=281444)
exten => _9./3005,n,Set(CALLERID(number)=281445)
exten => _9./3006,n,Set(CALLERID(number)=281446)


i believe these four should be on the same priority. asterisk dialplans 
will bork if the next priority doesnt match.


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