Re: [asterisk-users] Free CNAM

[Pascal Bruno]

If you can use curl, and can do some text parsing and know regular
expressions, you may be able to use this free CNAM service:
http://www.numberguru.com/ and integrate into your system.  This one appears
to have a more complete database.  When I tried my number, I have gotten my
full name, but when I use the FreeCNAM project below, I just get Florida.

On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally
wrote:

> I've been toying around with the idea of starting some kind of 'Open CNAM'
> project to destroy the current money hustle BS that dominates this industry.
>  The ever-growing FreeCNAM database may be a good starting point for such a
> project.
>
> I would also like to use Bitcoin (BTC) as the micropayment solution for
> user-requested updates.  Some nominal fee.
>
> If anyone wants to get involved, contact me.
>
>
>
>
> On 06/01/2011 07:51 AM, Skyler wrote:
>
>> Hi,
>>
>>  The junk in CNAM databases like "FLORIDA", "ONTARIO" etc. is IMO the
>> carrier's way to isolate their users and another excuse to charge more
>> money
>> for 'the better plan'. In the end, it's the carrier that inputs the info
>> so
>> if it shows "FLORIDA" with one database I can't see how any other database
>> would be different as the carrier is the only one that controls the
>> outbound
>> CID info. Calling me from POTS to snatch the CID will result in the same.
>>
>> ...unless there were a user friendly CNAM service, where info could be
>> updated by the end-user and queried freely by voip providers. I would
>> update
>> my cellular numbers for sure and know at least a dozen people that would
>> do
>> the same. Everyone is going VoIP so why not?
>>
>>  Talking about 'where's the money or angle'... here is one, vanity. Charge
>> $1/yr to a user per DID, if I don't renew then delete it and re-query the
>> original carrier.
>>
>>
>> S.
>>
>>
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Re: [asterisk-users] OK, I'm stumped

[Pascal Bruno]

That's how analog lines work. Asterisk do not know when the called is picked
up so it goes straight to the context execution.  You may want to try
setting callprogress=yes and answeronpolarity=yes on your chan dahdi conf
file as a work around, or switch to PRI



On Sun, May 16, 2010 at 3:38 PM, Adolphe Cher-aime wrote:

> Mi too I've experienced the same problem with my script. Dahdi answers the
> channel once Ami is running it's the same thing for call files . When using
> sip Chanel or skype channel it work as I wanted. I thank that analog fxo is
> the problem if automatic outgoing calls when you want the called party to
> answer first befor moving to the context extension.
>
> Adolphe Cher-aime
> From my Iphone
>
> On May 16, 2010, at 1:32 PM, Jose Flores Galicia 
> wrote:
>
> Maybe because I am closer to several customers which often make questions
> like yours.
>
> I can supposse you mean that the call is answered by the dahdi channel as
> soon as you set the originate command on AMI, I supposse you are using an
> FXO channel connected to your POTS line.
>
> Am I right?
>
> Jose Flores Galicia
> << floj...@gmail.com>>
> BriefCode && Code Based Training
>
>
> 2010/5/16 Bruce Ferrell < bferr...@baywinds.org>
>
>> I'm trying to make an AMI call.  I want to call a number, play an
>> announcement when the call is answered, then call a second number and
>> connect the two when the second call is answered.
>>
>> I an able to make a simple call to two numbers and connect them using
>> the manager API but playing the announcement has me beat.
>>
>> Suggestions anyone?
>>
>> Bruce Ferrell
>>
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Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server

[Pascal Bruno]

Why don't you copy the files to your asterisk box and play them from there?

-- Sent from my Android device

On Apr 16, 2010 5:03 PM, "Edwin Quijada"  wrote:




Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with ...
I did. But my problem is not conversion my problem is that I dont know how
play the file from windows server or copy this to asterisk without my AGI
continue and desyncronyze it.

Can you explain me exactly what did you do /?

Do you have something like this using AGI ?

I use sox with good results too in windows. The problem is when create the
file and convert it , how send to asterisk


Edwin Jaws

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[asterisk-users] Converting GSM calls to SIP

[Pascal Bruno]

I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me.  I would like to know
what is needed from the operator side to do this, what kind of material is
needed, or what can be done from their side to send SIP calls to  my server.

Thank you
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Re: [asterisk-users] Play an audio file from a remote host

[Pascal Bruno]

You might be able to do it using FastAGI

http://www.voip-info.org/wiki/view/Asterisk+FastAGI


On Mon, Mar 8, 2010 at 4:33 AM, Pham Quy  wrote:

> hi all,
>
> We going to implement a music service which enable user to playback a
> song by dialing to a service number.
>
> The problem is that the amount of data is huge so we have to plae it on
> an different server which is connected to the asterisk's via internet.
>
> Does asterisk support playing a audio file from an resource locate in a
> remote host?
>
>
> Please help,
>
> Thanks,
> Quyps
>
>
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Re: [asterisk-users] how to create a dummy call

[Pascal Bruno]

This may help you:

http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out



On Wed, Mar 3, 2010 at 11:20 PM, Pham Quy  wrote:

> Hi all,
>
> It maybe not clear that what i'm going to do.
> What i want to do is that enable user to call to a number then a
> background music will be played and he/she sing to mobilephone, the
> voice will be recorded and synchronized with the music.
>
> Any idea?
>
> There is an approach which using Monitor and Meetme application, it
> however need to throw an extra call to playing music, and this call
> should be thrown automatically by Asterisk.
>
> Again, any idea?
>
> Please help, thanks
> Quyps
>
> On Thu, 2010-03-04 at 10:37 +0700, Pham Quy wrote:
> > Hi all,
> >
> > What i'm going to do is that enable caller sing while playing a
> > background music (likes karaoke). My approach is using Monitor and
> > Meetme apps.Caller make a call to asterisk, asterisk join caller in to a
> > voice conference and create a dummy caller which will play music, then
> > Monitor app record both music and singer's voice.
> >
> > But i dont know how to create a dummy caller or throw a dummy call in
> > order to do above task.
> >
> > Any idea or comment is appreciated.
> >
> > Thanks
> > Quyps
>
>
>
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Re: [asterisk-users] IP Phone recommendation

[Pascal Bruno]

I would love to hear some inputs on Aastra and Snom IP phones.



On Wed, Feb 10, 2010 at 4:36 PM, Jeff LaCoursiere  wrote:

>
> On Wed, 10 Feb 2010, Tim Nelson wrote:
>
> > - "Gordon Henderson" 
> > >
> wrote:
> >> If not using PoE I'd suggest getting a few extra PSUs though - that's
> >> one
> >> area I have had a few issues with - but maybe it's just been the UK
> >> ones.
> >>
> >> Gordon
> >
> > The same can be said for the US versions. My experience has been it's not
> a case of 'if' the PSU will fail, but 'when'. In a past (less intelligent)
> life, I deployed a fair number of the GXP2020s and GXP2000s. There are not
> very many of them left that haven't completely died(the phone itself), and
> of those left, they've all had power supplies replaced.
> >
> > I cannot speak for the quality of the later devices from Grandstream.
> After being burned, it's a bit hard to look at them again when there are so
> many other quality devices available (think Polycom, Aastra, etc).
> >
> > --Tim
> >
>
> I haven't used any standard Grandstream IP phones, but I am *trying* to
> stabalize the new video phones they have come up with.  I have several
> GXV3000 and GXV3140s.  I got through central provisioning using their java
> based tool and for the most part these phones work, but have very odd
> bugs.  If left to itself for more than a few days the 3140 simply stops
> answering calls.  The 3000 has very odd DTMF issues - like doubling every
> digit pressed.  This is all fine and I know they are new products, but
> what is frustrating is Grandstream's lack of support.  The forums are next
> to useless, and the firmware releases are always "coming very soon".
>
> Then there are my horrid experiences with their FXO gateways.  Echo, bad
> audio in general, needing a reboot every few days, etc.  Again, support is
> non existant.
>
> So regardless of the quality of the latest phones, the company itself
> leaves a lot to be desired IMO.
>
> j
>
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Re: [asterisk-users] Problem with Portech MV-372

[Pascal Bruno]

I finally saw why it was doing it: In Mobile -> Settings -> SIP From field
there is 4 options:
Tel/User (Standard)
User/User (Standard)
Tel/Tel/ (Not Reg)
User/Tel (Not Reg)

when I choose any of the first two, I dont have this problem but when I use
the last two I have this problem.  At the same time, if I use the first two,
I am not getting the caller id of the person who called the sim, but in the
cdr I see the name of the extensions the gateway was registered too.

So what I had to do, is to set a fixed IP to the gateway and instead of
having host=dynamic I set host=ip_of_gateway.

This way the gateway does not have to register, and I can keep the settings
that passes the right caller id.  Another way would be to have asterisk read
another field for the caller id, because the number of the caller is
somewhere on the sip invite.


2009/11/27 Massimo Nuvoli 

> Pascal Bruno ha scritto:
> > Hi,
> >
> > I am experiencing a weird issue with my MV-372.
> >
> > Mobile1 & Mobile2 are both registered to my asterisk server, I am able
> > to use them for outgoing call with no problem, but when I call the sims
> > in my gateway, they are routed to the right context/extension/priority,
> > but as soon as I hangup, the sim unregistered from asterisk and tries to
> > register with my the callerid of the last incoming call as follows:
> >
> > Registration from '"mv372" 
> > 
> > <mailto:sip%3a%2b17546542...@77.29.9.16>>'
> failed for '97.26.196.2' - No
> > matching peer found
> >
> > and the registration fails since I dont have a peer created for
> +17546542334
> >
> > Anyone have an idea on how to go about fixing this?
>
> I am using the MV-372 (in and out) and dont have this problem.
>
> First: check if the device has the LATEST firmware, if not, upgrade.
>
> Second: send an email to the portech service. :-)
>
> In the past there was a lot of bug in the firmware of the MV372, and
> also buggy hardware release, but not now... so check also the hardware
> version (in the web interface -> firmware update -> top on the page).
>
> I think this is not "asterisk" issue.
>
> Bye.
>
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[asterisk-users] Problem with Portech MV-372

[Pascal Bruno]

Hi,

I am experiencing a weird issue with my MV-372.

Mobile1 & Mobile2 are both registered to my asterisk server, I am able to
use them for outgoing call with no problem, but when I call the sims in my
gateway, they are routed to the right context/extension/priority, but as
soon as I hangup, the sim unregistered from asterisk and tries to register
with my the callerid of the last incoming call as follows:

Registration from '"mv372"
>'
failed for '97.26.196.2' - No matching peer found

and the registration fails since I dont have a peer created for +17546542334

Anyone have an idea on how to go about fixing this?


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Re: [asterisk-users] Voicemail after hangup

[Pascal Bruno]

What are you trying to achieve here? The h extension is for when the  
channel hangs up. And if the caller hangs up how will he leave you a  
voicemail?


Sent from my iPod

On Nov 11, 2009, at 7:20 AM, Anahi Ludueña   
wrote:



Hi people, just a question:
Is it possible to execute Voicemail command in the h extension?  
(after hangup the channel).
Because if I put it before it, it works right, but if I put it  
there, it doesn't...

The log is:

-- Executing [...@cont-mine:1] NoOp("SIP/3005-096736a8", "End of  
cont-mine") in new stack
-- Executing [...@cont-mine:2] VoiceMail("SIP/3005-096736a8",  
"3003|su") in new stack
--  Playing '/var/spool/asterisk/voicemail/ 
default/3003/unavail' (language '')
  == Spawn extension (cont-mine, h, 2) exited non-zero on 'SIP/ 
3005-096736a8'


Thanks,


Charlas más divertidas con el nuevo Windows Live Messenger
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Re: [asterisk-users] installing

[Pascal Bruno]

Lol

Sent from my iPod

On Oct 27, 2009, at 6:59 AM, Alex Balashov   
wrote:

> aster...@opensourcesolution.in wrote:
>
>> installing asterisk
>
> I am intrigued by your ideas and would like to subscribe to your
> quarterly newsletter, as well as attend your biannual leadership  
> seminar.
>
> -- 
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
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Re: [asterisk-users] Best Firewall Suggestions?

[Pascal Bruno]

pfsense

On Tue, Oct 13, 2009 at 12:12 PM, Doug Lytle  wrote:

> David Wathen wrote:
> >
> > Hi,
> >
> > My customer has a outdated firewall that is also presenting a NAT
> > nightmare for getting the Asterisk server reachable from the internet.
> >
> > What firewalls work good with VOIP? I really want to steer away from
> > any ALG supported firewall. I just want a good firewall that works
> > well with Asterisk.
> >
>
> I'd suggest a Linux based firewall (pf or iptables) along with Firewall
> Builder:
>
> http://www.fwbuilder.org
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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Re: [asterisk-users] [SPAM] RE: dCAP Exam

[Pascal Bruno]

On Wed, Sep 16, 2009 at 2:37 PM, Zoa  wrote:

>
>  What if i send my twin brother to take the exam instead of me... ?
>
> z
>
>
If you think you cannot pass the test yourself, your twin wont be able to
pass it neither, he can be even worst than you

lol
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Re: [asterisk-users] dCAP Exam

[Pascal Bruno]

I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.

On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher wrote:
>
>> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
>> > Hmm...so by open book, that means access to the internet? Possible to
>> > get own notes ?
>>
>> Yes, you have access to the Internet, but your access is proxied, and the
>> administrator of the test can see everything that you access.  So it's
>> best
>> for you stick with only general guides and not look for crib notes.  If
>> your
>> test proctor believes you cheated, you fail.
>>
>> --
>> Tilghman Lesher
>> Digium, Inc. | Senior Software Developer
>> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
>> Check us out at: www.digium.com & www.asterisk.org
>>
>
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
> dump sites.
>
> Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
>
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Re: [asterisk-users] SIP and other phones other then local network

[Pascal Bruno]

For example if it was Alex to reply to that msg, i would feel bad for  
this guy, because Alex would make him feel like if he cannot do this  
by himself or use google to find that answer by himself, he does not  
belong to that list. He would never give him a chance and try to help  
him.

Sent from my iPod

On Sep 1, 2009, at 3:53 AM, Matt Riddell  wrote:

> On 1/09/09 7:48 PM, ABBAS SHAKEEL wrote:
>> Hello
>>
>> Please advice how can i configure a sip phone that is not on my local
>> network.  ie i have Xlite far some where in America and my Asterisk
>> server is at Sahara desert . Now how can i make a call to that sip  
>> phone?
>>
>>
>> Please advice what keywords to carry on??
>
> Search for Asterisk SIP NAT.
>
> Basically you'll need to port forward 5060 and the rtp ports
> (1-2 by default) to your Asterisk machine from the firewall.
>
> The outside person then registers to your machine (by using the  
> external
> address).
>
> -- 
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
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> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
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Re: [asterisk-users] Need help - CDR MySQL

[Pascal Bruno]

You have to fix the dependency issues, which means install the stuff  
you are missing that cdrmysql depends on so u can recompile it.

Sent from my iPod

On Aug 30, 2009, at 11:18 AM, Cyprus VoIP  wrote:

> Thanks. I found out that the module didn't load:
>
> [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
> Error loading module 'cdr_addon_mysql.so':
> /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
> object
> file: No such file or directory
>
> When I checked, I saw that it doesn't exist. It seems that when I
> installed the addons, I didn't realize that there were some issues to
> resolve first, as in the menuselect, I see that both  
> app_addon_sql_mysql
> and cdr_addon_mysql have dependencies problems.
>
> What should I do to resolve that?
>
> Thanks.
>
>  Original Message  
> Subject: Re: [asterisk-users] Need help - CDR MySQL
> From: Tilghman Lesher 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sunday, 30 August, 2009 17:17:59
>
>> On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
>>> Hello all,
>>>
>>> I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
>>> but
>>> without success.
>>>
>>> Is there a proper online manual that describes all the steps to  
>>> follow
>>> and debugging/monitoring information?
>>>
>>> When I type in the CLI "module show", cdr_addon_mysql.so is not  
>>> listed,
>>> although in modules.conf, I added the line "load =>  
>>> cdr_addon_mysql.so".
>>> I also tried "preload", but it didn't change anything.
>>>
>>> I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire
>>> necessary data for the mysql server is there.
>>>
>>> In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in  
>>> "[general]".
>>>
>>> I would appreciate any help I can get at this point, as I'm  
>>> clueless as
>>> to what can be wrong.
>>
>> Type:  'module load cdr_addon_mysql.so' and correct any errors you  
>> see.
>>
>
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Re: [asterisk-users] Asterisk Autodialer

[Pascal Bruno]

On Tue, Aug 25, 2009 at 2:52 PM, Alex Balashov wrote:

> With enough spiritual commitment, anything can be done;  you certainly
> *can* do it this way.  You can write a fairly sophisticated dialer in
> Bash, too.
>
> The issue is whether it is methodologically correct and qualitatively
> appropriate.  It is much easier to schedule calls and manage outcomes
> dynamically - such as highly granular agent stats informed by
> up-to-the-minute heuristics, or aggressive overdial ratio control - with
> real-time monitoring of both call initiation and call results via AMI.
>
> It's a question of ROI on your time.  You can do it however you want,
> especially if you're really motivated to avoid a particular type of
> development chore.


What you are saying makes sense, but I haven't used AMI for anything yet, so
I cannot comment on how easy/reliable/stable it is do work with in terms of
developing an autodialer, but I just did not agree when you said it was
definitely the way to go, because there is not one way to go.  The one I
did, using call files to dial was pretty reliable, and I you are able to
distribute the calls different servers using the interface.
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Re: [asterisk-users] Asterisk Autodialer

[Pascal Bruno]

On Tue, Aug 25, 2009 at 1:29 PM, Alex Balashov wrote:

> Sanjoy Rath wrote:
>
> > I would prefer to use AMI. Let me start looking into AMI. I would like
> > to include functionalities like upload numbers to call from an
> > interface, i want reports back numbers called, setup call time etc. Let
> > me look up AMI. Thanks Alex for the info.
>
> Yep.  If you need that level of detail, AMI is definitely the right
> approach.


I do not quite agree, I have developed a system exactly like that using call
files, and I do have an interface to upload the numbers to call, you can
setup call time, and the the delay to wait between each call.  To me it was
very straight forward and it works great, you can make a few thousands of
calls a day depending on how many lines/channels you have.
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Re: [asterisk-users] how to install asterisk

[Pascal Bruno]

http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu




On Fri, Aug 21, 2009 at 5:21 PM, trebaum  wrote:

> Start here, http://www.asterisk.org/support
>
> ~T
>
>
> On Aug 21, 2009, at 1:22 PM,  <
> aster...@opensourcesolution.in
>  > wrote:
>
> > hello friends,
> >
> > i have to configures asterisk n my hardware details are
> >
> >
> > O.S - Ubuntu 8.04 Lts
> >
> > Memory - 1 GB
> >
> > Proccessor- core 2 duo
> >
> > is any one having a good link or how to related asterisk.
> >
> > any help,support will be higly appreciated
> >
> > thx
> >
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Re: [asterisk-users] IPKall and FWD

[Pascal Bruno]

I have a DID from IPKall that is forwarded to my Asterisk box and I have
done no special configuration for SIP URI:

On IPKall you put the IP address of your asterisk box or the hostname then
is Sip phone number you put the did number they gave you and thats it.

On your asterisk box you set your peer in sip.conf then in you
extensions.conf, whatever you put in your sip phone number, that's what you
have to use as the extension.


On Thu, Aug 20, 2009 at 2:54 PM, randulo  wrote:

> On Thu, Aug 20, 2009 at 11:15 AM, SIP wrote:
> > IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one,
> > I'm not sure, but someone around has surely used it), etc, etc. There
> > are a lot of alternatives about.
>
> Sorry, I forgot to mention IdeaSIP.com which works great, I've had an
> account for years,  and of course your own Asterisk box as Gordon
> says. You need to be able to accept call via a SIP URI but that's easy
> enough to configure.
>
> We will be talking about this in relation the "Free DID" subject
> tomorrow on http://VUC.me
>
> /r
>
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Re: [asterisk-users] Skype for Asterisk???

[Pascal Bruno]

Lol but he has a good point and makes a lot of sense.  Never thought about
that strategy...


On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon
wrote:

> Michael Graves wrote:
> > Pricing is a very legitimate way to minimise support effort. It winnows
> > down the market size to a point where the company offering the goods
> > can sustain the projected per user support issues.
> >
> > You can always drop the price later on when you have a better handle on
> > the per user support issue.
> >
> > Michael
> >
> You make it sound like you're saying it's expensive because it doesn't
> work :-)
>
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Re: [asterisk-users] Play Fake ring in phpagi

[Pascal Bruno]

Why not record a ring tone, and playback the file? with $agi->streamfile???



On Tue, Aug 18, 2009 at 11:38 AM, Barton Fisher  wrote:

>
>  I'm going blind searching - maybe you know?
>>
>> During the execution of a script I want to play fake ring to caller. Both
>> of these examples complain of missing option:
>>
>> $agi->exec("Ringing");
>> $agi->exec("Playtones ring");
>>
>
>
>> Notice: Undefined variable: options in
>> /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326
>>
>> Warning: Missing argument 2 for AGI::exec(), called in
>> /var/lib/asterisk/agi-bin/dax-ivr.agi on line 156 and defined in
>> /var/lib/asterisk/agi-bin/includes/phpagi.php on line 323
>>
>>  I changed to $agi->exec("Playtones ring",""); -  no error message but not
> sure that correct
>
>> Any ideas what to put for missing option?
>>
>> TIA
>>
>> Bart
>>
>
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[asterisk-users] Skype for Asterisk???

[Pascal Bruno]

Not sure if anybody noticed, but it seems like Skype For Asterisk is out.

$66 per channels, pretty pricey

http://store.digium.com/productview.php?product_code=1SFA0001
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Re: [asterisk-users] Asterisk + CDRTool

[Pascal Bruno]

Did you get CDRTool to work with Asterisk or Areski's CDR Stats?



On Fri, Aug 14, 2009 at 10:20 AM, harry R  wrote:

> Hi
>
> I just solve my problem today. Just a package on redhat that I need
> install.
>
> H.
>
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Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?

[Pascal Bruno]

Where you able to compile DAHDI in a virtual environment?  How about skype
for asterisk?  Has anyone tried that in a virtual environment?  Seems like
to register the license, digium tool is looking for a connection on eth0,
and in a virtual environment I see the name as vnet0 or vnet1.  At least
that what I see on godaddy's virtual servers.


On Fri, Aug 7, 2009 at 12:08 PM, Tarek Sawah  wrote:

>
> been testing with Sun VirtualBox  and i managed more than 30 extensions on
> a 2GHz Dual core machine with 1 GB ram for the VBOX.. just not running
> recodring or encoding .. things went well
>
> --
> AHD Tarek Sawah
> 
> > Date: Fri, 7 Aug 2009 08:47:03 -0700
> > From: jlama...@gmail.com
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Asterisk in VMWare, how does it perform and
> what is the limit?
> >
> > Hi,
> > I'm coming up with ideas about building a cluster of asterisk servers,
> > and am exploring the virtualization option.
> > I'm curious to know some real-world data about how many extensions a
> > VMWare install on good hardware could support.
> > I've seen stories about how the hypervisor timeslicing can wreak havoc
> > on call quality at some point.
> > Is this really the case? If so, what's a feasible extension limit? 20?
> 50? 100?
> >
> > Any information would be great.
> >
> > Thanks.
> >
> > -- James
> >
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>
> _
> Windows Live™: Keep your life in sync.
>
> http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

Well I think thats what the problem was, I dont have it named as eth0.  So
if your NIC is not labeled eth0 you cannot use skypeforasterisk???  Why cant
it just scan you nic handles?  Can someone point me to where I can change
the NIC name in the source file or something???



On Sun, Aug 2, 2009 at 1:05 PM, Steve Totaro  wrote:

> On Sun, Aug 2, 2009 at 12:13 PM, randulo  wrote:
>
>> On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote:
>> > So what do you think I can do to register my license? I am running
>> > Asterisk 1.6.10 on CentOS 5.
>>
>> >>> Could not generate Host-ID.
>> >>> Make sure that you have eth0 enabled.
>>
>> The MAC is used in the scheme to register and it looks like it can't
>> be read for some reason. There must be a direct channel to Digium for
>> the support of this kind, though. Have you tried contacting them?
>>
>> [waits for John Todd to chime in here...]
>>
>
>
> Is eth0 enabled?  Is it named eth0?
>
> What does ifconfig eth0 tell you?
>
> I have seen many Dell servers where the two NICs are labeled eth1 and eth2
> or whatever, but in Linux, they are backwards.  Eth2 show up as eth0 and
> eth1 shows up as eth1 in Linux.
>
> Wasted a good half hour to forty five minutes trying to figure out why I
> couldn't get the network up.
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

So what do you think I can do to register my license? I am running  
Asterisk 1.6.10 on CentOS 5.

Sent from my iPod

On Aug 2, 2009, at 3:49 AM, Thomas Kenyon   
wrote:

> Pascal Bruno wrote:
>> Unfortunately for me, I cannot register my license.  Kept saying:
>>
>> Could not generate Host-ID.
>> Make sure that you have eth0 enabled.
>>
>> Any help would be appreciated
>>
> It uses the same licensing scheme as the G.729 licenses (so as soon as
> you need to upgrade the machine, or set up LACP or VPN or any other  
> type
> of virtual interface or in the case of G.729 you change the codec to a
> newer version {since you've upgraded to a new version of asterisk that
> doesn't support older ones} that doesn't support the old name for the
> codec, you need to re-register).
>
> Or as in your case, it doesn't like the names of the network  
> interfaces.
>
> It's all a total PITA.
>
> Fwiw, the Skype channel driver stopped working on my machine a while
> ago. I never did track down the cause.
>
> When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
> start and 1 skypewatcher service starts.
>
> If I start it manually after asterisk has started, usually asterisk
> segfaults, (not always).
>
> Although Sometimes it starts up properly but can't log anyone in,  
> Either
> the user is stated as Logged Out or Connection Error, usually if I  
> type
> skype show users I get the following error message:
>
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
>
> (Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
> skypeforasterisk-1.6.1_0.9.10-x86_64)
>
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Re: [asterisk-users] Converting sound files

[Pascal Bruno]

On linux you can use Sox. Google it and resd the documentation to see  
how you can convert files from the command line. On windows you can  
use Switch by NCH Software. Download the trial then you can pay a  
small fee if you want to keep it.

Sent from my iPod

On Aug 2, 2009, at 10:30 AM, "Christian"  wrote:

> Hi all,
> I have a set of sound files that are recorded in 16 Bit 44.1 KHz  
> stereo and I want to convert them into 16 bit 8000 KHz mono so that  
> i can use them in Asterisk.
> What is the best way of doing that?
> Many thanks,
> Christian
>
>
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

Unfortunately for me, I cannot register my license.  Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.

On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning  wrote:

>
> Nice job.  It worked right away for me with my 10 channel trial license.
> Asterisk 1.4.26
>
> I'm already building a dtmf access menu to bridge to my SIP world :-)
>
> As much I hate Skype for being a closed system, it would make the ultimate
> "remote" Asterisk extension as Skype drills through so many firewalls that
> block SIP and IAX and just about everything else.
>
>
>
>
>
> On Thu, Jul 30, 2009 at 1:50 PM, John Todd  wrote:
>
>>
>> I know many of you have been waiting for this for a while, so I'll
>> keep this short:  The Skype for Asterisk Public Beta is now available
>> on the Digium store.
>>
>> We are pleased to announce the open beta of Skype For Asterisk is
>> ready to begin and we look forward to you participation. To obtain
>> your copy of the software, please visit Digium’s web store and
>> purchase (for zero dollars) the Skype For Asterisk product. The web
>> store does require a Digium.com account, which can be set up during
>> the purchase process if you don’t already have one.
>> Once the web store process is complete, you will be e-mailed your
>> license key and directions on where to download Skype For Asterisk
>> beta software.
>>
>> This is a "time-expiring" beta - the software will stop working on
>> August 31.  The download is also currently time-limited - it will be
>> available until August 7 on our website.  After the 31st, you would
>> need to have purchased a license for the SfA software (sorry, no
>> pricing that I can give you right now - that will be a separate
>> announcement.  I'm just the community guy - I have no idea about
>> pricing or commercial contracts or the like, so please wait until
>> that's been announced as I will find out about the same time as you
>> do. :-)
>>
>> Trial "purchase" page:
>>   http://store.digium.com/productview.php?product_code=804-00019
>>
>> JT
>>
>> ---
>> John Todd   
>> email:jt...@digium.com
>> Digium, Inc. | Asterisk Open Source Community Director
>> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
>> direct: +1-256-428-6083 http://www.digium.com/
>>
>>
>>
>>
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Re: [asterisk-users] Skype for Asterisk: Public Beta available

[Pascal Bruno]

Unfortunately for me, I cannot register my license.  Kept saying:

Could not generate Host-ID.
Make sure that you have eth0 enabled.

Any help would be appreciated


On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning  wrote:

>
> Nice job.  It worked right away for me with my 10 channel trial license.
> Asterisk 1.4.26
>
> I'm already building a dtmf access menu to bridge to my SIP world :-)
>
> As much I hate Skype for being a closed system, it would make the ultimate
> "remote" Asterisk extension as Skype drills through so many firewalls that
> block SIP and IAX and just about everything else.
>
>
>
>
>
> On Thu, Jul 30, 2009 at 1:50 PM, John Todd  wrote:
>
>>
>> I know many of you have been waiting for this for a while, so I'll
>> keep this short:  The Skype for Asterisk Public Beta is now available
>> on the Digium store.
>>
>> We are pleased to announce the open beta of Skype For Asterisk is
>> ready to begin and we look forward to you participation. To obtain
>> your copy of the software, please visit Digium’s web store and
>> purchase (for zero dollars) the Skype For Asterisk product. The web
>> store does require a Digium.com account, which can be set up during
>> the purchase process if you don’t already have one.
>> Once the web store process is complete, you will be e-mailed your
>> license key and directions on where to download Skype For Asterisk
>> beta software.
>>
>> This is a "time-expiring" beta - the software will stop working on
>> August 31.  The download is also currently time-limited - it will be
>> available until August 7 on our website.  After the 31st, you would
>> need to have purchased a license for the SfA software (sorry, no
>> pricing that I can give you right now - that will be a separate
>> announcement.  I'm just the community guy - I have no idea about
>> pricing or commercial contracts or the like, so please wait until
>> that's been announced as I will find out about the same time as you
>> do. :-)
>>
>> Trial "purchase" page:
>>   http://store.digium.com/productview.php?product_code=804-00019
>>
>> JT
>>
>> ---
>> John Todd   
>> email:jt...@digium.com
>> Digium, Inc. | Asterisk Open Source Community Director
>> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
>> direct: +1-256-428-6083 http://www.digium.com/
>>
>>
>>
>>
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Re: [asterisk-users] Maximum number of concurrent calls

[Pascal Bruno]

This question has been asked thousands of time on this list.  You mat want
to search the archive, but to sum it all, there is no limit as far as calls
on an Asterisk server.  It all depends on your server's specs, and how it is
setup.  A celeron processor with 256Mb Ram could handle a fews calls where a
Quad Core with 4Gb Ram would handle way more.  So it all depends on the
platform and the way you configure you Asterisk server.  Also you can
cluster Asterisk servers making its capacity pretty much unlimited, if you
know what you doing nd if you doing it right.



On Sat, Aug 1, 2009 at 6:46 AM, Emrah  wrote:

> Hi,
>
> I remember reading that Asterisk allows only 100 simultaneous calls. Is
> that correct?
> If it is so, how is it possible to have a conference call with more then
> 100 users? I think I read here that some people managed to have 500
> people in a conf room...
> Or, how do I increase this limit? Is it as easy as changing a value in a
> config.h?
>
> Thanks for your help,
> Emrah
>
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Re: [asterisk-users] using asterisk on a shared line

[Pascal Bruno]

That's right, they say it is a PBX because it is mostly used as  such, but
it is more than just a PBX.  Some people use it as a VoiceMail tool or to
handle just conference, some use it to add functionalities to other legacy
PBX systems.  Calling cards applications for example, a plain PBX wont be
able to do that.  Thats why I dont usually refer to it as a PBX.


Pascal,
>
> I agree with you that Asterisk is a telephony applications toolkit, and not
> a simple answering machine.  However, Asterisk IS a PBX.
>
> The term "answering machine" in the context of this thread implies a device
> that has only basic answering functionality.  Since Asterisk is capable of
> so much more than this basic functionality, I encouraged the OP to use it
> full time, rather than as an adjunct device.
>
> First line at:
> http://www.asterisk.org/
>
> "Asterisk is the world's leading open source PBX, telephony engine, and
> telephony applications toolkit."
>
> Sincerely,
> Trevor Hammonds
>
>
>
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Re: [asterisk-users] using asterisk on a shared line

[Pascal Bruno]

Just a little clarification for people refering to Asterisk as a PBX  
and not an Answering Machine:

In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk  
is a Telephony Toolkit. You can choose to use it as a PBX or an  
Answering Machine or both or even in some case as a something  
different than a PBX or Answering Machine. You should know that  
already, so this is just a reminder :-)

Sent from my iPod

On Jul 23, 2009, at 8:34 PM, "Trevor Hammonds"   
wrote:

> Bill Lovett wrote:
>>
>> Can Asterisk be configured to hang up if another phone picks up?
>>
>> I'm a bit lost as far as terminology goes, but here's my setup. At
>> home, I have asterisk answering calls from the pstn and sending them
>> through to a sip extension or voicemail. All that is working fine.
>>
>> The box running Asterisk isn't on 24/7 so I have a secondary phone
>> connected to the line as well. If Asterisk is not running, I can
>> answer an incoming call from that phone. If asterisk is running, I  
>> can
>> answer the call from a sip extension.
>>
>> Can I have it both ways? Can Asterisk back off if the secondary phone
>> answers the call? Currently, if a call comes in and I answer it from
>> the secondary phone Asterisk will continue to ring the sip extension
>> and eventually drop into voicemail.
>
> Asterisk is a PBX, not an answering machine, so I would advise  
> against this.
> It would be best to have Asterisk handle the phone line exclusively,  
> 24/7.
> However, with that said, it is possible to accomplish what you are  
> asking.
>
> Placing a telephone privacy/exclusion adapter on the line cord into  
> Asterisk
> will cut off the phone line whenever a parallel telephone on the  
> same line
> is picked up.  This means that the instant you pick up any other  
> phone on
> the line, it would cut off the line to Asterisk.
>
> Radio Shack used to sell a couple varieties of these.  One was a two- 
> way
> adapter with one side for "phone" and the other "answering  
> machine".  You do
> not need to plug anything into the "phone" side for the device to  
> work.  The
> second device was just an inline exclusion device.  I was unable to  
> find
> these at Radio Shack's website.  However, I found something similar  
> at the
> following URLs:
>
> (See SER2A, SER2D, and SER3P at Sandman.com)
> http://www.sandman.com/lineshar.html
>
> http://www.trianglecables.com/telanmacorph.html
>
> http://www.iec-usa.com/cgi-bin/iec/COM9928
>
> http://www.iec-usa.com/cgi-bin/iec/COM0006
>
> Good luck!
>
> Sincerely,
> Trevor Hammonds
>
>
>
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Re: [asterisk-users] Asterisk, SQL Database Update

[Pascal Bruno]

If you have asterisk addons installed you can use the mysql  
applications to make queries. I find it to be very easy if you know  
how to do select and insert queries and understand the basic mechanism  
of the dialplan. Other than that, you may want to hire someone to do it.

Sent from my iPod

On May 25, 2009, at 5:03 PM, Edwin Quijada  
 wrote:

>
>
>>
>> Thanks for your helpful reply.
>>
>>
>>
>> I am not so good in coding.
>>
>>
>>
>> simply all i need is as follow
>>
>>
>>
>> When a call comes, goes into an IVR, and then depending on the  
>> entry option
>>
>> it will connect to a remote SQL Database, to check the pre-existed  
>> data,
>>
>> and in the end of the IVR the caller will enter an option that will  
>> need to be written in the SQL Database.
>>
>>
>>
>> Can you please give me a general scenrio how this will be achieved.
>>
>> and which files that i will need to modify.
>>
>
> I think that if you are not good coding you will have a few problems.
> Maybe, the best solution 4u is hire external to do that. It is simple
> but just in dialplan it is so difficult with AGI it is so easy but
> you dont want coding.
>
>
>
> *---*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-809-849-8087
> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera  
> de lo comun"
> *---*
>
> _
> Windows Live Hotmail now works up to 70% faster.
> http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster_112008
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Re: [asterisk-users] Open source SIP client

[Pascal Bruno]

It seems like a few people including me DID understand what Dhaval meant, or
maybe some people used they common sense and their intelligence to
understand what somebody who's english is not the primary language wanted to
say and put some effort to guide or help someone in the community getting to
the right direction instead of trying to put him down.

I think a few others need to consider investigating more deeply the basic
mechanics of understanding written English, or should themselves research
what some collections of syllables intend to convey.  I also think if they
were that good, why not provided some english tutoring instead of putting
people down.

Good luck in you research Dhaval!




On Mon, May 18, 2009 at 9:46 AM, Scott Gifford wrote:

> DHAVAL INDRODIYA  writes:
>
> > can anybody help me to give Opensource SIP client information which
> > can be modified as per our requirment
>
> Hello Dhaval,
>
> We have tried several open-source SIP phones on Linux.  We have had
> the best luck with Twinkle Phone:
>
>http://www.xs4all.nl/~mfnboer/twinkle/index.html
>
> It has lots of hooks where you can stick your own scripts to modify
> its behavior.  We also had pretty good luck with SFLphone:
>
>http://www.sflphone.org/
>
> There is a list of open source clients on voip-info that includes
> these two.  It might be a good starting point:
>
>http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software
>
> Good luck!
>
> Scott.
>
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Re: [asterisk-users] Execute after hangup

[Pascal Bruno]

You can use the extension h

exten => h,1,app()
exten => h,n,app()
.

On Mon, Apr 20, 2009 at 10:31 AM, Michael wrote:

> What is the syntax to progress with a dial plan after hangup please?
>
> Michael
>
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Re: [asterisk-users] AT&T PRI Install - What is outpulsed?

[Pascal Bruno]

I believe she is refering to how she's going to send you your incoming calls
(on your DIDs) for example:
10 digits: 972-453-2345
7 digits: 453-2345
4 digits: 2345

so you know how to expect your incoming calls and configure your
extensions.conf accordingly.




On Fri, Mar 27, 2009 at 10:06 AM, Dave Fullerton <
dfullertaster...@shorelinecontainer.com> wrote:

> Hey All,
>
> AT&T is installing a PRI in a couple weeks and while I've been doing
> homework on PRI's for the last few weeks there's something I'm still
> confused about. After being asked how many digits I wanted them to send
> us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked
> her what that meant and all I got was the question repeated. Do any of
> you have any idea what she was referring to? Is this ANI? Outgoing
> Caller ID? Something else?
>
> While I've done many POTS line setups, this is my first PRI install, so
> I'd also welcome any "make sure you do this", "read this first" or "AT&T
> always messes this up so..." tips.
>
> Thanks
>
> -Dave
>
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Re: [asterisk-users] I need a country, state, city database

[Pascal Bruno]

You may want to check this link

http://www.geodatasource.com/cities-free.html

it may help you



On Sun, Mar 22, 2009 at 8:14 PM, Cary Fitch  wrote:

>  I don’t, but it out to be “out there”.  We needed a list of all (valid)
> bank routing numbers for a check writing program and a former associate
> found that for free.
>
>
>
> I suggest you look in the direction of the US Department of Commerce.  They
> have to have a list of what you want, and the basic information is pretty
> static.
>
>
>
> (We know all the states, ;-), there are no new counties either, and not
> much changes in towns.)
>
>
>
> Map makers, GPS manufacturers, Census, Post Office, FCC, 1000 other govt.
> agencies, UPS,. FedEx etc all use such a list.
>
>
>
> I would almost think Google search could come up with such a list.
>
>
>
> Cary Fitch
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
> *Sent:* Sunday, March 22, 2009 2:12 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] I need a country, state, city database
>
>
>
> I need a country, state, city database for a web application.
>
>
>
> Anyone have a free version they can email (or drop.io) for me?
>
>
>
> Looking for something like this at $197 but may as well ask in case you
> know of a free source.
>
> http://www.globixdata.com/pop.cfm?db=world&v1=l&v2=s&v3=a&pricing=99
>
>
>
>
>
>
>
>
>
> Regards,
>
> Dean Collins
> Cognation Inc
> d...@cognation.net
> +1-212-203-4357   New York
> +61-2-9016-5642   (Sydney in-dial).
> +44-20-3129-6001 (London in-dial).
>
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

Thanks for your help

  Don’t really know the answer, but these are “givens”:
>
>1. your phone is (most likely) in the same area code as the asterisk
>installation
>
> My phone has a different area code than the asterisk installation.  The
asterisk box is in FL and I can call a number in MN but not the 201 or many
others

>
>1.
>2. NY is most likely not in the same area code.
>
> I agree but I could call a MN cell phone for example which works all the
time

>
>1.
>2. Even though the T1 is a dedicated digital service, the code that
>handles all of this is/was written to process calls from analog sources for
>backwards compatibility and therefore would have the timing issue handlers
>in place even though they don’t apply.
>
>
>
> My research revealed that you might use an exception to stop this, but I
> didn’t really find a good example.  You could check viop-info.org or
> whirlpool to see what they say.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Friday, March 20, 2009 9:39 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> I still find it weird as even if it is a switch timing problem.  Because
> when is it calling my phone *all the time *and that other area code it *never
> *calls it.  Does that mean asterisk always complete my number in a certain
> time frame, and the other number no?  Also I get the progress code 127
> exactly after i move my call file to the outgoing folder, there is no delay,
> I get it tthe same time I move the move.
>
>
>
> And also why the call goes through when I put SIP/whatever in the
> callerid? Does that mean asterisk get to complete the call in the time
> frame?
>
> On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas  wrote:
>
> You can also do a set variable in the call file.  I don’t really know how
> to do that, but you can probably find the command and syntax on
> voip-info.org.
>
> The reason it works on certain numbers has to do with switch timing.  If *
> can complete the call within a certain time frame, all is well.  If not, the
> 127 thing will bite you.
>
> You would think we were past that type of thing, but I suppose not.
>
>
>
> Another thing you might try is changing the 60 to 90 or so on your original
> call file.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Thursday, March 19, 2009 4:42 PM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> I dont want to change it within my extensions.conf, because I have many
> dids, and change them on the fly according to the call i am making.  I have
> a web interface where I fill a form that gets the number I am calling, the
> caller id and context to go etc...
>
>
>
> I dont want to keep editing extensions.conf and reload, I want to do it
> directly in the call file.
>
>
>
> What I dont understand is WHY it works on certain numbers and not all.
>  That is a problem, it is not normal.
>
>
>
>
>
> On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas  wrote:
>
> GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the
> trick
>
>
> -Original Message-----
> From: asterisk-users-boun...@lists.digium.com
>
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Thursday, March 19, 2009 3:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 problem (call using a .call file)
>
> Pascal Bruno wrote:
> > Also very strange, when in my call file I change the callerid line to
> > SIP/whatever like Danny said, the call go through, but I dont want
> > that, because when I do so, it is displaying the main number on my T1
> > account as caller id and I dont want that, I want to display one of my
> > other DID as callerid.
>
>
> Then change your caller-id within your dialplan, not the callfile.
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

I still find it weird as even if it is a switch timing problem.  Because
when is it calling my phone *all the time *and that other area code it *never
*calls it.  Does that mean asterisk always complete my number in a certain
time frame, and the other number no?  Also I get the progress code 127
exactly after i move my call file to the outgoing folder, there is no delay,
I get it tthe same time I move the move.

And also why the call goes through when I put SIP/whatever in the
callerid? Does that mean asterisk get to complete the call in the time
frame?

On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas  wrote:

>  You can also do a set variable in the call file.  I don’t really know how
> to do that, but you can probably find the command and syntax on
> voip-info.org.
>
> The reason it works on certain numbers has to do with switch timing.  If *
> can complete the call within a certain time frame, all is well.  If not, the
> 127 thing will bite you.
>
> You would think we were past that type of thing, but I suppose not.
>
>
>
> Another thing you might try is changing the 60 to 90 or so on your original
> call file.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Thursday, March 19, 2009 4:42 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> I dont want to change it within my extensions.conf, because I have many
> dids, and change them on the fly according to the call i am making.  I have
> a web interface where I fill a form that gets the number I am calling, the
> caller id and context to go etc...
>
>
>
> I dont want to keep editing extensions.conf and reload, I want to do it
> directly in the call file.
>
>
>
> What I dont understand is WHY it works on certain numbers and not all.
>  That is a problem, it is not normal.
>
>
>
>
>
> On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas  wrote:
>
> GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the
> trick
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
>
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Thursday, March 19, 2009 3:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 problem (call using a .call file)
>
> Pascal Bruno wrote:
> > Also very strange, when in my call file I change the callerid line to
> > SIP/whatever like Danny said, the call go through, but I dont want
> > that, because when I do so, it is displaying the main number on my T1
> > account as caller id and I dont want that, I want to display one of my
> > other DID as callerid.
>
>
> Then change your caller-id within your dialplan, not the callfile.
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

I dont want to change it within my extensions.conf, because I have many
dids, and change them on the fly according to the call i am making.  I have
a web interface where I fill a form that gets the number I am calling, the
caller id and context to go etc...
I dont want to keep editing extensions.conf and reload, I want to do it
directly in the call file.

What I dont understand is WHY it works on certain numbers and not all.  That
is a problem, it is not normal.



On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas  wrote:

> GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the
> trick
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Thursday, March 19, 2009 3:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 problem (call using a .call file)
>
> Pascal Bruno wrote:
> > Also very strange, when in my call file I change the callerid line to
> > SIP/whatever like Danny said, the call go through, but I dont want
> > that, because when I do so, it is displaying the main number on my T1
> > account as caller id and I dont want that, I want to display one of my
> > other DID as callerid.
>
>
> Then change your caller-id within your dialplan, not the callfile.
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
> ___
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

Also very strange, when in my call file I change the callerid line to
SIP/whatever like Danny said, the call go through, but I dont want that,
because when I do so, it is displaying the main number on my T1 account as
caller id and I dont want that, I want to display one of my other DID as
callerid.




On Thu, Mar 19, 2009 at 4:23 PM, Pascal Bruno  wrote:

> Here is what I get from the console with the call file:
>
> -- Attempting call on DAHDI/g1/1201XXX for s...@fortest:1 (Retry 1)
> -- Requested transfer capability: 0x00 - SPEECH
> -- PROGRESS with cause code 127 received
> [Mar 19 16:12:47] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
> since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
> [Mar 19 16:12:52] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
> since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
> [Mar 19 16:12:57] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
> since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
> [Mar 19 16:13:02] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
> since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
> -- Hungup 'DAHDI/1-1'
>
> And here is what I get from using my analog phone:
>
>  -- Starting simple switch on 'DAHDI/32-1'
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '1'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '1' on DAHDI/32-1
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '1' on DAHDI/32-1
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '2'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '2' on DAHDI/32-1
> [Mar 19 16:23:50] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '2' on DAHDI/32-1
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '0'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '0' on DAHDI/32-1
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '0' on DAHDI/32-1
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '1'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '1' on DAHDI/32-1
> [Mar 19 16:23:51] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '1' on DAHDI/32-1
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '4' on DAHDI/32-1
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '4' on DAHDI/32-1
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '5' on DAHDI/32-1
> [Mar 19 16:23:52] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '5' on DAHDI/32-1
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '8' on DAHDI/32-1
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '8' on DAHDI/32-1
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '3' on DAHDI/32-1
> [Mar 19 16:23:53] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '3' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '1' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '1' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:54] DTMF[11266]: channel.c:

Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

 DAHDI/1-1 is proceeding passing it to DAHDI/32-1
-- DAHDI/1-1 is making progress passing it to DAHDI/32-1
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered DAHDI/32-1
-- Native bridging DAHDI/32-1 and DAHDI/1-1
-- Hungup 'DAHDI/1-1'


Call is fine with the phone, but does not go through with .call file






On Thu, Mar 19, 2009 at 11:47 AM, Danny Nicholas  wrote:

>  Try this call file – replace XXX with your number and YYY with a valid
> SIP exten on your system
>
>
>
> Channel: DAHDI/g1/1XX
> Callerid:  SIP/YYY
>
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Thursday, March 19, 2009 9:22 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> Here is what my extensions.conf file has:
>
>
>
> exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _NXXNXX,n,Hangup()
>
>
>
> exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _1NXXNXX,n,Hangup()
>
>
>
> Using the phone, I can dial any numbers succesfully.
>
>
>
> And here is my call file:
>
>
>
> Channel: DAHDI/g1/1XX
> Callerid: XX
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
> Context: test
> Extension: s
> Priority: 1
>
>
>
> with the call file I can dial my cellphone which begin with 754XXX
>
> but when I call my friend's cellphone from new york which is 201XXX i
> get progress code 127 as follows
>
>
>
> -- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
> -- Requested transfer capability: 0x00 - SPEECH
> -- PROGRESS with cause code 127 received
>
>
>
> I tried with the prefix 1 and without the prefix 1 it is always the same
> thing, but with the handset I dial my phone and my friend's phone
> succesfully with and without the 1
>
>
>
>
>
>
>
> On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas  wrote:
>
> Please paste the call file content (with the number ’ed of course) and
> the Dial section from extensions.conf.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Wednesday, March 18, 2009 6:24 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> This has to be a bug, because I dont know what else to try here
>
>
>
>
>
> On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno  wrote:
>
> Nope, I always dial 1 + 10 digits for all my numbers.  It works on all
> numbers when I am using my phone (Analogue or IP) but when I do it using a
> .call file it does not work on some numbers mostly.  That is the weirdest
> thing I have ever seen.  I tried different codecs in the call file, I still
> get the PROGRESS with cause code 127
>
>
>
>
>
>
>
>
>
> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg 
> wrote:
>
> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno  wrote:
> > I have a weird problem with call using my T1 card.  I can make calls fine
> > using my analog and IP phones, but when I try to initiate a call using a
> > .call file, I get the following error
> >  -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- PROGRESS with cause code 127 received
> > it happens on certain numbers I dial, but if I dial that same number with
> an
> > ip or analog phone that use the T1 channel, the call is going through
> > normally.
> > Anybody knows why?
>
> Are you doing anything silly with prefixing or short-circuit dialing?
>
> in other words..
>
> You dial 8 for an outside line, then 1+10 digits
> and you're forgetting to do that for some numbers?
>
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>
>
>
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

Here is what my extensions.conf file has:

exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _NXXNXX,n,Hangup()

exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1NXXNXX,n,Hangup()

Using the phone, I can dial any numbers succesfully.

And here is my call file:

Channel: DAHDI/g1/1XX
Callerid: XX
MaxRetries: 1
RetryTime: 5
WaitTime: 60
Context: test
Extension: s
Priority: 1

with the call file I can dial my cellphone which begin with 754XXX
but when I call my friend's cellphone from new york which is 201XXX i
get progress code 127 as follows

-- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received

I tried with the prefix 1 and without the prefix 1 it is always the same
thing, but with the handset I dial my phone and my friend's phone
succesfully with and without the 1




On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas  wrote:

>  Please paste the call file content (with the number ’ed of course)
> and the Dial section from extensions.conf.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Wednesday, March 18, 2009 6:24 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> This has to be a bug, because I dont know what else to try here
>
>
>
>
>
> On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno  wrote:
>
> Nope, I always dial 1 + 10 digits for all my numbers.  It works on all
> numbers when I am using my phone (Analogue or IP) but when I do it using a
> .call file it does not work on some numbers mostly.  That is the weirdest
> thing I have ever seen.  I tried different codecs in the call file, I still
> get the PROGRESS with cause code 127
>
>
>
>
>
>
>
>
>
> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg 
> wrote:
>
> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno  wrote:
> > I have a weird problem with call using my T1 card.  I can make calls fine
> > using my analog and IP phones, but when I try to initiate a call using a
> > .call file, I get the following error
> >  -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- PROGRESS with cause code 127 received
> > it happens on certain numbers I dial, but if I dial that same number with
> an
> > ip or analog phone that use the T1 channel, the call is going through
> > normally.
> > Anybody knows why?
>
> Are you doing anything silly with prefixing or short-circuit dialing?
>
> in other words..
>
> You dial 8 for an outside line, then 1+10 digits
> and you're forgetting to do that for some numbers?
>
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>
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> To UNSUBSCRIBE or update options visit:
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>
>
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

This has to be a bug, because I dont know what else to try here


On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno  wrote:

> Nope, I always dial 1 + 10 digits for all my numbers.  It works on all
> numbers when I am using my phone (Analogue or IP) but when I do it using a
> .call file it does not work on some numbers mostly.  That is the weirdest
> thing I have ever seen.  I tried different codecs in the call file, I still
> get the PROGRESS with cause code 127
>
>
>
>
>
> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg wrote:
>
>> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno  wrote:
>> > I have a weird problem with call using my T1 card.  I can make calls
>> fine
>> > using my analog and IP phones, but when I try to initiate a call using a
>> > .call file, I get the following error
>> >  -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
>> > -- Requested transfer capability: 0x00 - SPEECH
>> > -- PROGRESS with cause code 127 received
>> > it happens on certain numbers I dial, but if I dial that same number
>> with an
>> > ip or analog phone that use the T1 channel, the call is going through
>> > normally.
>> > Anybody knows why?
>>
>> Are you doing anything silly with prefixing or short-circuit dialing?
>>
>> in other words..
>>
>> You dial 8 for an outside line, then 1+10 digits
>> and you're forgetting to do that for some numbers?
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

Nope, I always dial 1 + 10 digits for all my numbers.  It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call file it does not work on some numbers mostly.  That is the weirdest
thing I have ever seen.  I tried different codecs in the call file, I still
get the PROGRESS with cause code 127





On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg wrote:

> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno  wrote:
> > I have a weird problem with call using my T1 card.  I can make calls fine
> > using my analog and IP phones, but when I try to initiate a call using a
> > .call file, I get the following error
> >  -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- PROGRESS with cause code 127 received
> > it happens on certain numbers I dial, but if I dial that same number with
> an
> > ip or analog phone that use the T1 channel, the call is going through
> > normally.
> > Anybody knows why?
>
> Are you doing anything silly with prefixing or short-circuit dialing?
>
> in other words..
>
> You dial 8 for an outside line, then 1+10 digits
> and you're forgetting to do that for some numbers?
>
> ___
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> To UNSUBSCRIBE or update options visit:
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[asterisk-users] T1 problem (call using a .call file)

[Pascal Bruno]

I have a weird problem with call using my T1 card.  I can make calls fine
using my analog and IP phones, but when I try to initiate a call using a
.call file, I get the following error
 -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received

it happens on certain numbers I dial, but if I dial that same number with an
ip or analog phone that use the T1 channel, the call is going through
normally.

Anybody knows why?

My call file looks like this:

Channel: DAHDI/g1/1XX
Callerid: XX
MaxRetries: 1
RetryTime: 5
WaitTime: 60
Context: test
Extension: s
Priority: 1
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Re: [asterisk-users] Help Inbound number

[Pascal Bruno]

Your sip.conf should look like this
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default

and extensions.conf

[default]
exten = 246463,1,Dial(SIP/8003)

you must also have a sip user for 8003 in your sip.conf like
[8003]
type=friend
username=XX
secret=XX
context=outgoing

And dont forget to do a sip reload and dialplan reload



On Mon, Mar 16, 2009 at 6:23 PM, Bayardo Sanchez
wrote:

> The inbound was working well suddenly stopped working I want all calls made
> to the number  should answer the extension 8003
>
>
> On Mon, Mar 16, 2009 at 3:49 PM, Danny Nicholas  wrote:
>
>>  Just to read this right – you are trying to take an inbound call from
>> 888xxx and transfer it to your sip extension 8003?
>>
>>
>>
>> If so,
>>
>> Are you able to make internal calls to 8003?
>>
>> Can you transfer other calls to 8003 (exten => s,1,Dial(SIP/8003)
>> ) ?
>>
>>
>>  --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
>> *Sent:* Monday, March 16, 2009 4:38 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Help Inbound number
>>
>>
>>
>> nothing the problem persitem
>>
>> On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee  wrote:
>>
>> are you sure calls from this provider are going to context 'default' ?
>>
>> sip.conf
>> [procall]
>> type=peer
>> username=XX
>> secret=XX
>> context=default
>>
>> 2009/3/16 Bayardo Sanchez 
>>
>> i create inbound number but i calling and send this error:
>>
>> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
>> Call from '101396_procall' to extension '246463' rejected because
>> extension not found.
>>
>> but the extensin existed
>>
>> --
>> Bayardo Sánchez García
>> Web Developer - Internet Portals - Asterisk Support - Windows Server
>> Support - Proxi Support
>> E-mail: bayardo.sanc...@gmail.com
>> Linux User: #418392
>> America Central - Managua, NI (505) 249-2853 -  4886876
>> IM msn messenger: bjsanch...@hotmail.com
>> Skype: bayardo.sanchez
>> This email is intended solely for the person or organization to which it
>> is addressed. It may contain privileged and confidential information. If you
>> are not the intended recipient, you are prohibited from copying, disclosing
>> or distributing this email or its contents (as it may be unlawful for you to
>> do so) or taking any action in reliance on it. If you have received this
>> email by mistake, please delete it. All e-mail sent to this address will be
>> received by B.S. Solution e-mail system and is subject to archiving and
>> review by someone other than the recipient.
>>
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>>
>>
>> --
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>> Web Developer - Internet Portals - Asterisk Support - Windows Server
>> Support - Proxi Support
>> E-mail: bayardo.sanc...@gmail.com
>> Linux User: #418392
>> America Central - Managua, NI (505) 249-2853 -  4886876
>> IM msn messenger: bjsanch...@hotmail.com
>> Skype: bayardo.sanchez
>> This email is intended solely for the person or organization to which it
>> is addressed. It may contain privileged and confidential information. If you
>> are not the intended recipient, you are prohibited from copying, disclosing
>> or distributing this email or its contents (as it may be unlawful for you to
>> do so) or taking any action in reliance on it. If you have received this
>> email by mistake, please delete it. All e-mail sent to this address will be
>> received by B.S. Solution e-mail system and is subject to archiving and
>> review by someone other than the recipient.
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>
>
>
> --
> Bayardo Sánchez García
> Web Developer - Internet Portals - Asterisk Support - Windows Server
> Support - Proxi Support
> E-mail: bayardo.sanc...@gmail.com
> Linux User: #418392
> America Central - Managua, NI (505) 249-2853 -  4886876
> IM msn messenger: bjsanch...@hotmail.com
> Skype: bayardo.sanchez
> This email is intended solely for the person or organization to which it is
> addressed. It may contain privileged and confidential information. If you
> are not the intended recipient, you are prohibited from copying, disclo

Re: [asterisk-users] Help Inbound number

[Pascal Bruno]

Do you have an extension set for 246463 in your extensions.conf?





On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez
wrote:

> i create inbound number but i calling and send this error:
>
> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
> Call from '101396_procall' to extension '246463' rejected because
> extension not found.
>
> but the extensin existed
>
> --
> Bayardo Sánchez García
> Web Developer - Internet Portals - Asterisk Support - Windows Server
> Support - Proxi Support
> E-mail: bayardo.sanc...@gmail.com
> Linux User: #418392
> America Central - Managua, NI (505) 249-2853 -  4886876
> IM msn messenger: bjsanch...@hotmail.com
> Skype: bayardo.sanchez
> This email is intended solely for the person or organization to which it is
> addressed. It may contain privileged and confidential information. If you
> are not the intended recipient, you are prohibited from copying, disclosing
> or distributing this email or its contents (as it may be unlawful for you to
> do so) or taking any action in reliance on it. If you have received this
> email by mistake, please delete it. All e-mail sent to this address will be
> received by B.S. Solution e-mail system and is subject to archiving and
> review by someone other than the recipient.
>
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Re: [asterisk-users] getting free Did number for asterisk

[Pascal Bruno]

check ipkall.com

On Sat, Mar 14, 2009 at 12:46 PM, Meftah Tayeb wrote:

> hello
> please ho to get a free did number for my asterisk ?
> also, is it pocible to assign it to a group of extentions ?
> thanks!
>
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Re: [asterisk-users] No reply to our critical packet

[Pascal Bruno]

I have the same situation.  My scenario is weird:

I have a DID with IPkall that points to my asterisk server, and I have it
play a message with Playback()  after about 20 seconds call drops and give
me the same message you get: "no reply to our critical packet"

BUT

I have a DID from Vitelity, and that one works fine no drops.

I have no idea why.



On Fri, Mar 13, 2009 at 12:37 PM, Roman Odaisky  wrote:

> On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote:
>
> > Next Step would be to check/update the firmware on your phones or router.
>
> I don’t think the router is to blame, it does deliver all the packets. And
> there are no hardware phones, only numerous software SIP clients.
>
> Which (GNU/Linux) software clients are known to have maximum compatibility
> with Asterisk?
>
> --
> TIA
> Roman.
>
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Re: [asterisk-users] dialstatus through a call file

[Pascal Bruno]

My call file was calling an AGI application, and from with the AGI, I could
not get the DIALSTATUS,  I will try to send it to the dialplan first, then
call my AGI from the dialplan and see what happen.

Thanks for your help


On Tue, Feb 3, 2009 at 3:35 AM, Johansson Olle E  wrote:

>
> 3 feb 2009 kl. 04.33 skrev Ex Vito:
>
> > On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno 
> > wrote:
> >> Is it possible to retrieve the DIALSTATUS variable when placing
> >> call through
> >> a call file.  This variable is set when using the Dial()
> >> application from
> >> the dialplan, but I am using a call file for my current application
> >> and need
> >> to get the dialstatus.
> >
> >  Your call file will initiate actions defined in the dialplan and
> > certainly after
> >  the triggered Dial the DIALSTATUS will be available to the dialplan.
> >
> >  Now the question is: "where" do you want to retreive the DIALSTATUS
> > to ?
> >
> >  If back to the OS environment (a file ?) you will need to have your
> > dialplan
> >  do it for you, maybe via System(echo ${DIALSTATUS} > /tmp/file) or
> > something...
> >  (NOTE: i'm not sure of the syntax of the application... check it
> > with "core show
> >  application System" on the CLI)
>
> A call file can either direct a call to an application or to a
> specific extension.
> Instead of sending the call to Dial, create an extension where you grab
> the dialstatus after the call and store it away somewhere for the
> application
> to retrieve.
>
> Or convert your app to AMI and you will get it all.
>
> /O
>
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Re: [asterisk-users] Module res_odbc is not loading

[Pascal Bruno]

Actually I installed them after, so do you recommend I recompile asterisk?If
I do so, I wont loose my current configuration files right?

On Tue, Jan 27, 2009 at 6:27 PM, Philipp Kempgen
wrote:

> Pascal Bruno schrieb:
> > I have remove the comment defor res_odbc.so and res_config_odbc.so in my
> > modules.conf, but the module is still not loading
> >
> > when I do:
> >
> > module show like odbc
> > I have o module returned
> >
> > anybody knows why?
>
> Did you install unixodbc and unixodbc-dev before compiling Asterisk?
>
>
>   Philipp Kempgen
>
> --
> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
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[asterisk-users] Module res_odbc is not loading

[Pascal Bruno]

Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading

when I do:

module show like odbc
I have o module returned

anybody knows why?
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[asterisk-users] dialstatus through a call file

[Pascal Bruno]

Hello,
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file.  This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and need
to get the dialstatus.

Thank you.
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[asterisk-users] Help with cdr_odbc

[Pascal Bruno]

I have having a hard time setting the cdr with cdr_odbc.  Below is all my
conf file related to it, let me know what I am doing wrong.  Thank you.

*cdr.conf*
[general]
enable=yes

[csv]
usegmtime=yes; log date/time in GMT.  Default is "no"
loguniqueid=yes  ; log uniqueid.  Default is "no"
loguserfield=yes ; log user field.  Default is "no"

*cdr_odbc.conf**
*[global]
dsn=MySQL-asterisk
loguniqueid=yes
dispositionstring=yes
table=cdr
usegmtime=no
username=myuser
password=mypass

*modules.conf*
[modules]
autoload=yes
preload => res_odbc.so
preload => res_config_odbc.so
preload => func_strings.so
noload => pbx_gtkconsole.so
load => res_musiconhold.so
load => cdr_odbc.so
noload => chan_alsa.so

*res_odbc.conf
*[ENV]
[asterisk]
enabled => yes
dsn => MySQL-asterisk
username => myuser
password => mypass
pre-connect => yes*

odbc.ini*
[MySQL-asterisk]
Description = Asterisk MySQL ODBC
Driver = MySQL
Socket   = /var/run/mysqld/mysqld.sock
Server   = localhost
User   = myuser
Password= mypass
Database= asterisk
Option  = 3
#Port   =

*odbcinst.ini
*[MySQL]
Description = MySQL driver
Driver  = /usr/lib/odbc/libmyodbc.so
Setup   = /usr/lib/odbc/libodbcmyS.so
FileUsage   = 1



I have my table "cdr" in the asterisk database, I am using mysql and
asterisk 1.6.0.3.  The mysql user has all priviledges granted on host %(any)

Any help would be appreciated
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Re: [asterisk-users] Logging outgoing calls

[Pascal Bruno]

That is a good idea too, where would I configure asterisk to log the channel
status on that custom field?





On Sat, Jan 24, 2009 at 8:27 AM, David fire  wrote:

> and what about add a custome field or setup a variable on outgoing calls
> and use the common cdr and then filtering by that field.
> David
>
> 2009/1/24 Tilghman Lesher 
>
> On Friday 23 January 2009 18:22:16 Pascal Bruno wrote:
>> > Is it possible to log just the outgoing calls using cdr_odbc into a
>> custom
>> > mysql database table?
>> > my table will look like this:
>> >  
>> >
>> > |  call_status   |
>> > |-- --|
>> > | · id   |
>> > | · destination  |
>> > | · status |
>> > ||
>> >
>> > I just need to store the destination number and the status of the
>> channel
>> > for example BUSY, UNAVAILABLE etc...
>>
>> Yes, if you install the cdr_adaptive_odbc backport.  See the sample config
>> file for more information.
>>
>> Web:  http://svncommunity.digium.com/view/tilghman/branches/1.4
>> SVN:  http://svncommunity.digium.com/svn/tilghman/branches/1.4
>>
>> --
>> Tilghman
>>
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>
>
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
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[asterisk-users] Logging outgoing calls

[Pascal Bruno]

Is it possible to log just the outgoing calls using cdr_odbc into a custom
mysql database table?
my table will look like this:
 
|  call_status   |
|-- --|
| · id   |
| · destination  |
| · status |
||

I just need to store the destination number and the status of the channel
for example BUSY, UNAVAILABLE etc...
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Re: [asterisk-users] Problem with TDM808

[Pascal Bruno]

With:

callprogress=yes
and
progzone=us

it works fine for not, not 100% percent because in some calls,it takes like
3-4 seconds before executing dialplan, which is not bad not to say normal.
But most calls are ok.

And when I tried with

answeronpolarityswitch=yes

it doesnt do dialplan at all, like the call was never picked up.

Thanks for your help!




On Tue, Jan 20, 2009 at 6:53 PM, D Tucny  wrote:

> If your provider provides any signalling to indicate answer, such as a
> polarity reversal, this could be detected easily...
>
> ; Use a polarity reversal to mark when a outgoing call is answered by the
> ; remote party.
> ;
> ;answeronpolarityswitch=yes
>
> This isn't very common though... alternatively, there is the 'HIGHLY
> EXPERIMENTAL' call progress detection...
>
> ; On trunk interfaces (FXS) it can be useful to attempt to follow the
> progress
> ; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
> ; progress attempts to determine answer, busy, and ringing on phone lines.
> ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
> ; so don't count on it being very accurate.
> ;
> ; Few zones are supported at the time of this writing, but may be selected
> ; with "progzone".
> ;
> ; progzone also affects the pattern used for buzydetect (unless
> ; busypattern is set explicitly). The possible values are:
> ;   us (default)
> ;   ca (alias for 'us')
> ;   cr (Costa Rica)
> ;   br (Brazil, alias for 'cr')
> ;   uk
> ;
> ; This feature can also easily detect false hangups. The symptoms of this
> is
> ; being disconnected in the middle of a call for no reason.
> ;
> ;callprogress=yes
> ;progzone=uk
>
> Obviously far from ideal, and at least, where I am, unworkable due to the
> way that all the telcos have got into providing musical ringing...
>
> The only real solution is to go digital...
>
> d
>
>
> 2009/1/21 Pascal Bruno 
>
> Is there any way of going around this???  Any tricks, configuration hacks??
>>
>>
>>
>>
>>
>> On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith  wrote:
>>
>>> On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
>>> > I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
>>> > 1.6.3.  When I try making a call with a .call file, the call goes
>>> > straight to the dialplan and start executing the dialplan even before
>>> > the called party has pick up.  Anybody knows why by any chance?
>>>
>>> That's not a problem with the TDM800 card... it's just a side-effect of
>>> analog signaling.  For analog calls, the central office doesn't give any
>>> type of signal when the far end has answered the call, so Asterisk has
>>> no way of knowing when that happens. For that reason, Asterisk
>>> immediately treats any outgoing analog call as having been answered.
>>>
>>> --
>>> Jared Smith
>>> Digium, Inc. | Training Manager
>>>
>>>
>>>
>>>
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Re: [asterisk-users] Problem with TDM808

[Pascal Bruno]

Is there any way of going around this???  Any tricks, configuration hacks??




On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith  wrote:

> On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
> > I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
> > 1.6.3.  When I try making a call with a .call file, the call goes
> > straight to the dialplan and start executing the dialplan even before
> > the called party has pick up.  Anybody knows why by any chance?
>
> That's not a problem with the TDM800 card... it's just a side-effect of
> analog signaling.  For analog calls, the central office doesn't give any
> type of signal when the far end has answered the call, so Asterisk has
> no way of knowing when that happens. For that reason, Asterisk
> immediately treats any outgoing analog call as having been answered.
>
> --
> Jared Smith
> Digium, Inc. | Training Manager
>
>
>
>
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[asterisk-users] Problem with TDM808

[Pascal Bruno]

Dear List,

I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3.
When I try making a call with a .call file, the call goes straight to the
dialplan and start executing the dialplan even before the called party has
pick up.  Anybody knows why by any chance?

Any help would be appreciated.
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Re: [asterisk-users] Text messaging and Asterisk

[Pascal Bruno]

Is it possible for asterisk to send sms through a GSM gateway, tor example
the Portech MV-37X?
If yes, any examples of configurations would be really apreciated.



On Tue, Oct 14, 2008 at 11:13 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

> The most flexible way but will require a bit of work and scales SMS modem
> per SMS per second.
>
> Install kannel and configure it to work with your SMS modem (many cell
> phones work just fine for sending and receiving).  It does not have to go on
> the asterisk box, just a box you can hit with HTTP or HTTPs.
>
> Make sure you have lynx installed
>
> In your Asterisk dialplan use system(lynx
> http://ipofyoursmsserverusernamepasswordnumbermessage) that is not the
> exact syntax but it is all documented but everything in the SMS in encoded
> in the URL.
>
> With five T-mobile phones, I can send five a second, it seems linear, it
> may be possible to increase throughput, I just got it working and left it at
> that.
>
> Messages queue until there is an available modem, sent in order.
>
> PLUS it is MUCH cheaper (at least in the US) than these aggregators.  With
> a T-Mobile family plan, 1,000 voice minutes, and unlimited SMS runs me about
> $135/mo.  I guess it depends on volume of SMS you are sending.
>
> Thanks,
> Steve Totaro
>
>
> On Tue, Oct 14, 2008 at 11:46 PM, C. Savinovich <
> c.savinov...@itntelecom.com> wrote:
>
>>
>>  Thanks, excellent point.  Furthermore, a google search on fastsms.conf
>> yielded the existence of a couple of 'Asterisk SMS gateways'..wow
>>
>> CS
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Drew Gibson
>> Sent: Tuesday, October 14, 2008 2:22 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Text messaging and Asterisk
>>
>> C. Savinovich wrote:
>> >   Can somebody please give a pointer to a complete neophyte (like me) on
>> > text messaging, what product can I use to send and automatic text
>> message
>> to
>> > a cell phone from within the asterisk dialplan? (the part of the
>> dialplan
>> I
>> > have down, the part of the text message no)
>> >
>> > Thanks
>> > C. Savinovich
>> >
>> >
>>
>> I don't use it but on my Asterisk 1.4 slug there was a file
>> /etc/asterisk/fastsms.conf which had info about connecting to SMS
>> services for about 4c per txt.
>>
>> regards,
>>
>> Drew
>>
>>
>> --
>> Drew Gibson
>>
>> Systems Administrator
>> OANDA Corporation
>> www.oanda.com
>>
>>
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>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Sorry for bothering you, but I got it, I just had to put # in callnum!



On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno  wrote:

> I want to dial out using the sim card.  What I did, I have used the SIP
> channel ex:
>
> Channel: SIP/thenum...@mv378
>
> It shows the called is being made in the dialplan, but the number I have
> entered does not dial, it just goes straight to the specified dialplan
> extensions.
>
> Then what I did, in the Lan to Mobile Table, I put * in url and the number
> I wanted to dial in call num, then the call was made to that number using
> the sim card properly.
>
> I was wondering if I cannot supply the number to be dialed using an
> asterisk call file, or do I have to put that number in the Lan to Mobile
> table.
>
> Any help would be appreciated.
>
> Thanks
>
>
>
>
>
> On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno  wrote:
>
>> Marco,
>>
>> The configs work fine for me.  I can receive calls with no problem.  Now,
>> were you able to dial using the sim card?  I cant figure out how I can do it
>> since asterisk doesnt have a channel to place call through the portech
>> gateway.
>>
>>
>>
>>
>>
>> On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno wrote:
>>
>>> Thank you!, I will try that in a few hours and let you know what happens.
>>>
>>>
>>>
>>> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini 
>>> wrote:
>>>
>>>>
>>>>
>>>> Pascal Bruno wrote:
>>>>
>>>> Thanks for your reply!
>>>>
>>>> Can you tell me what you have in your Portech configuration settings
>>>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>>>> pretty similar to yours but still cant register.
>>>>
>>>>
>>>>
>>>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini 
>>>> wrote:
>>>>
>>>>> Emmanuel Pascal Bruno wrote:
>>>>>
>>>>>  Has anyone been able to configure portech's mv-378 gateway with
>>>>> asterisk?
>>>>>
>>>>> I did the configuration as per the manual but it does not work.
>>>>>
>>>>> My server sees the portech gateway, but when the gateway is trying to
>>>>> register to my server it fails.  It says peer is not suppose to register.
>>>>>
>>>>> The gateway and the asterisk box are on two different location (two
>>>>> network, 2 differrent IP address).
>>>>>
>>>>> I would appreciate any kind of tutorial or advice on how to make it
>>>>> work.
>>>>>
>>>>> Thanks
>>>>>
>>>>> --
>>>>>
>>>>> ___
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>
>>>>> Hi,
>>>>> I've an installation working with Portech MV-370. I'm supposing it's
>>>>> quite similar to what you have. If it could be useful to you, this is my
>>>>> sip.conf configuration file.
>>>>>
>>>>> [GSMGtw1]
>>>>> type=friend
>>>>> context=from-gsm
>>>>> host=dynamic; we have a DHCP assigned address
>>>>> secret=reallyverysecret
>>>>> nat=no  ; there is not NAT between phone and
>>>>> Asterisk
>>>>> canreinvite=no
>>>>> dtmfmode=INFO
>>>>> insecure=invite ; required to overcome authentication
>>>>> problems in incoming calls
>>>>> call-limit=1   ; permit only 1 outgoing call at a
>>>>> time
>>>>> disallow=all
>>>>> allow=ulaw
>>>>> allow=alaw
>>>>> allow=gsm
>>>>> qualify=500
>>>>>
>>>>> I remember that I've found a bug on the firmware that prevents to the
>>>>> unit to register correctly on my asterisk box unless I'm using the raw IP
>>>>> address instead of the name of the asterisk box. I remember something 
>>>>> wrong
>>>>> in cryptogra

Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

I want to dial out using the sim card.  What I did, I have used the SIP
channel ex:

Channel: SIP/thenum...@mv378

It shows the called is being made in the dialplan, but the number I have
entered does not dial, it just goes straight to the specified dialplan
extensions.

Then what I did, in the Lan to Mobile Table, I put * in url and the number I
wanted to dial in call num, then the call was made to that number using the
sim card properly.

I was wondering if I cannot supply the number to be dialed using an asterisk
call file, or do I have to put that number in the Lan to Mobile table.

Any help would be appreciated.

Thanks





On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno  wrote:

> Marco,
>
> The configs work fine for me.  I can receive calls with no problem.  Now,
> were you able to dial using the sim card?  I cant figure out how I can do it
> since asterisk doesnt have a channel to place call through the portech
> gateway.
>
>
>
>
>
> On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno  wrote:
>
>> Thank you!, I will try that in a few hours and let you know what happens.
>>
>>
>>
>> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini 
>> wrote:
>>
>>>
>>>
>>> Pascal Bruno wrote:
>>>
>>> Thanks for your reply!
>>>
>>> Can you tell me what you have in your Portech configuration settings
>>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>>> pretty similar to yours but still cant register.
>>>
>>>
>>>
>>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini 
>>> wrote:
>>>
>>>> Emmanuel Pascal Bruno wrote:
>>>>
>>>>  Has anyone been able to configure portech's mv-378 gateway with
>>>> asterisk?
>>>>
>>>> I did the configuration as per the manual but it does not work.
>>>>
>>>> My server sees the portech gateway, but when the gateway is trying to
>>>> register to my server it fails.  It says peer is not suppose to register.
>>>>
>>>> The gateway and the asterisk box are on two different location (two
>>>> network, 2 differrent IP address).
>>>>
>>>> I would appreciate any kind of tutorial or advice on how to make it
>>>> work.
>>>>
>>>> Thanks
>>>>
>>>> --
>>>>
>>>> ___
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>
>>>> Hi,
>>>> I've an installation working with Portech MV-370. I'm supposing it's
>>>> quite similar to what you have. If it could be useful to you, this is my
>>>> sip.conf configuration file.
>>>>
>>>> [GSMGtw1]
>>>> type=friend
>>>> context=from-gsm
>>>> host=dynamic; we have a DHCP assigned address
>>>> secret=reallyverysecret
>>>> nat=no  ; there is not NAT between phone and
>>>> Asterisk
>>>> canreinvite=no
>>>> dtmfmode=INFO
>>>> insecure=invite ; required to overcome authentication
>>>> problems in incoming calls
>>>> call-limit=1   ; permit only 1 outgoing call at a
>>>> time
>>>> disallow=all
>>>> allow=ulaw
>>>> allow=alaw
>>>> allow=gsm
>>>> qualify=500
>>>>
>>>> I remember that I've found a bug on the firmware that prevents to the
>>>> unit to register correctly on my asterisk box unless I'm using the raw IP
>>>> address instead of the name of the asterisk box. I remember something wrong
>>>> in cryptography chiper/dechiper based on realm... So, if you have problems,
>>>> let's try to specify the asterisk raw IP address in the Portech.
>>>>
>>>> Best regards,
>>>> Marco Signorini.
>>>>
>>>>
>>>>
>>> Hi,
>>>
>>> I don't know if the problem could be in the Mobile to Lan or Lan to
>>> Mobile settings because these  settings are related on how calls coming
>>> from/to mobile are routed.  I didn't use the Portech routing features at all
>>> because I need a simple GSM gateway to

Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Marco,

The configs work fine for me.  I can receive calls with no problem.  Now,
were you able to dial using the sim card?  I cant figure out how I can do it
since asterisk doesnt have a channel to place call through the portech
gateway.




On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno  wrote:

> Thank you!, I will try that in a few hours and let you know what happens.
>
>
>
> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini wrote:
>
>>
>>
>> Pascal Bruno wrote:
>>
>> Thanks for your reply!
>>
>> Can you tell me what you have in your Portech configuration settings
>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>> pretty similar to yours but still cant register.
>>
>>
>>
>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:
>>
>>> Emmanuel Pascal Bruno wrote:
>>>
>>>  Has anyone been able to configure portech's mv-378 gateway with
>>> asterisk?
>>>
>>> I did the configuration as per the manual but it does not work.
>>>
>>> My server sees the portech gateway, but when the gateway is trying to
>>> register to my server it fails.  It says peer is not suppose to register.
>>>
>>> The gateway and the asterisk box are on two different location (two
>>> network, 2 differrent IP address).
>>>
>>> I would appreciate any kind of tutorial or advice on how to make it work.
>>>
>>> Thanks
>>>
>>> --
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>> Hi,
>>> I've an installation working with Portech MV-370. I'm supposing it's
>>> quite similar to what you have. If it could be useful to you, this is my
>>> sip.conf configuration file.
>>>
>>> [GSMGtw1]
>>> type=friend
>>> context=from-gsm
>>> host=dynamic; we have a DHCP assigned address
>>> secret=reallyverysecret
>>> nat=no  ; there is not NAT between phone and
>>> Asterisk
>>> canreinvite=no
>>> dtmfmode=INFO
>>> insecure=invite ; required to overcome authentication
>>> problems in incoming calls
>>> call-limit=1   ; permit only 1 outgoing call at a
>>> time
>>> disallow=all
>>> allow=ulaw
>>> allow=alaw
>>> allow=gsm
>>> qualify=500
>>>
>>> I remember that I've found a bug on the firmware that prevents to the
>>> unit to register correctly on my asterisk box unless I'm using the raw IP
>>> address instead of the name of the asterisk box. I remember something wrong
>>> in cryptography chiper/dechiper based on realm... So, if you have problems,
>>> let's try to specify the asterisk raw IP address in the Portech.
>>>
>>> Best regards,
>>> Marco Signorini.
>>>
>>>
>>>
>> Hi,
>>
>> I don't know if the problem could be in the Mobile to Lan or Lan to Mobile
>> settings because these  settings are related on how calls coming from/to
>> mobile are routed.  I didn't use the Portech routing features at all because
>> I need a simple GSM gateway to/from the asterisk box.
>> For this reason:
>> 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where 
>> "mob" is the extension I've generated in the asterisk box under the
>> context where the Portech operates;
>> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>>
>> I think the most relevant parameters for your problem are under the
>> "Service Domain" menu option (assuming that the firmware you have is similar
>> to what I've). On this menu I've compiled the 1st Realm (as I've only one
>> account) like that:
>>
>> UserName: GSMGtw1
>> RegisterName: GSMGtw1
>> RegisterPassword: reallyverysecret
>> Domain Server: 192.168.0.5
>> Proxy Server: 192.168.0.5
>>
>> Pay attention that, having specified the Domain Server with the raw IP
>> address, asterisk needs to be able to authenticate peers associated to that.
>> For this reason I've set:
>>
>> domain=192.168.0.5
>>
>> on sip.conf [general] section (remember to issue a sip reload from
>> asterisk cli).
>>
>> Hope this helps!
>>
>>
>> Best regards.
>> Marco Signorini
>>
>>
>>
>> 
>> Marco Signorini
>> INGEGNI Tech S.r.l.
>> http://www.ingegnitech.com
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Thank you!, I will try that in a few hours and let you know what happens.



On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini wrote:

>
>
> Pascal Bruno wrote:
>
> Thanks for your reply!
>
> Can you tell me what you have in your Portech configuration settings
> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
> pretty similar to yours but still cant register.
>
>
>
> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:
>
>> Emmanuel Pascal Bruno wrote:
>>
>>  Has anyone been able to configure portech's mv-378 gateway with
>> asterisk?
>>
>> I did the configuration as per the manual but it does not work.
>>
>> My server sees the portech gateway, but when the gateway is trying to
>> register to my server it fails.  It says peer is not suppose to register.
>>
>> The gateway and the asterisk box are on two different location (two
>> network, 2 differrent IP address).
>>
>> I would appreciate any kind of tutorial or advice on how to make it work.
>>
>> Thanks
>>
>> --
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> Hi,
>> I've an installation working with Portech MV-370. I'm supposing it's quite
>> similar to what you have. If it could be useful to you, this is my sip.conf
>> configuration file.
>>
>> [GSMGtw1]
>> type=friend
>> context=from-gsm
>> host=dynamic; we have a DHCP assigned address
>> secret=reallyverysecret
>> nat=no  ; there is not NAT between phone and
>> Asterisk
>> canreinvite=no
>> dtmfmode=INFO
>> insecure=invite ; required to overcome authentication
>> problems in incoming calls
>> call-limit=1   ; permit only 1 outgoing call at a time
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> qualify=500
>>
>> I remember that I've found a bug on the firmware that prevents to the unit
>> to register correctly on my asterisk box unless I'm using the raw IP address
>> instead of the name of the asterisk box. I remember something wrong in
>> cryptography chiper/dechiper based on realm... So, if you have problems,
>> let's try to specify the asterisk raw IP address in the Portech.
>>
>> Best regards,
>> Marco Signorini.
>>
>>
>>
> Hi,
>
> I don't know if the problem could be in the Mobile to Lan or Lan to Mobile
> settings because these  settings are related on how calls coming from/to
> mobile are routed.  I didn't use the Portech routing features at all because
> I need a simple GSM gateway to/from the asterisk box.
> For this reason:
> 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5 where
> "mob" is the extension I've generated in the asterisk box under the context
> where the Portech operates;
> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>
> I think the most relevant parameters for your problem are under the
> "Service Domain" menu option (assuming that the firmware you have is similar
> to what I've). On this menu I've compiled the 1st Realm (as I've only one
> account) like that:
>
> UserName: GSMGtw1
> RegisterName: GSMGtw1
> RegisterPassword: reallyverysecret
> Domain Server: 192.168.0.5
> Proxy Server: 192.168.0.5
>
> Pay attention that, having specified the Domain Server with the raw IP
> address, asterisk needs to be able to authenticate peers associated to that.
> For this reason I've set:
>
> domain=192.168.0.5
>
> on sip.conf [general] section (remember to issue a sip reload from asterisk
> cli).
>
> Hope this helps!
>
>
> Best regards.
> Marco Signorini
>
>
>
> 
> Marco Signorini
> INGEGNI Tech S.r.l.
> http://www.ingegnitech.com
>
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Thanks for your reply!

Can you tell me what you have in your Portech configuration settings (Mobile
to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is pretty
similar to yours but still cant register.



On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:

> Emmanuel Pascal Bruno wrote:
>
>   Has anyone been able to configure portech's mv-378 gateway with
> asterisk?
>
> I did the configuration as per the manual but it does not work.
>
> My server sees the portech gateway, but when the gateway is trying to
> register to my server it fails.  It says peer is not suppose to register.
>
> The gateway and the asterisk box are on two different location (two
> network, 2 differrent IP address).
>
> I would appreciate any kind of tutorial or advice on how to make it work.
>
> Thanks
>
> --
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi,
> I've an installation working with Portech MV-370. I'm supposing it's quite
> similar to what you have. If it could be useful to you, this is my sip.conf
> configuration file.
>
> [GSMGtw1]
> type=friend
> context=from-gsm
> host=dynamic; we have a DHCP assigned address
> secret=reallyverysecret
> nat=no  ; there is not NAT between phone and
> Asterisk
> canreinvite=no
> dtmfmode=INFO
> insecure=invite ; required to overcome authentication
> problems in incoming calls
> call-limit=1   ; permit only 1 outgoing call at a time
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> qualify=500
>
> I remember that I've found a bug on the firmware that prevents to the unit
> to register correctly on my asterisk box unless I'm using the raw IP address
> instead of the name of the asterisk box. I remember something wrong in
> cryptography chiper/dechiper based on realm... So, if you have problems,
> let's try to specify the asterisk raw IP address in the Portech.
>
> Best regards,
> Marco Signorini.
>
>
> ___
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Portech MV-378 with Asterisk

[Emmanuel Pascal Bruno]

Has anyone been able to configure portech's mv-378 gateway with asterisk?

I did the configuration as per the manual but it does not work.

My server sees the portech gateway, but when the gateway is trying to
register to my server it fails.  It says peer is not suppose to register.

The gateway and the asterisk box are on two different location (two network,
2 differrent IP address).

I would appreciate any kind of tutorial or advice on how to make it work.

Thanks
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Re: [asterisk-users] Recommend Wireless IP Phone

[Emmanuel Pascal Bruno]

The latest Nokia phones come with a SIP client and I like them.



On Wed, Nov 5, 2008 at 10:56 PM, Pedram M <[EMAIL PROTECTED]> wrote:

> Any recommendations on good wireless SIP phones?
>
> Thanks,
> Pedram
>
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Re: [asterisk-users] Call problems

[Emmanuel Pascal Bruno]

I have tried that too with no results





On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:

> Emmanuel Pascal Bruno wrote:
> > I have turned off firewall on the linux box, I have turned off
> > firewall on the router I still have the same problem :-(
>
> Disabling firewalls is almost certainly going to ensure the problem
> persists.  You need to ensure that all SIP and RTP ports are
> port-forwarded from your firewall to your Asterisk box.
>
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Re: [asterisk-users] Call problems

[Emmanuel Pascal Bruno]

I have turned off firewall on the linux box, I have turned off firewall on
the router I still have the same problem :-(



On Sat, Nov 1, 2008 at 1:22 PM, Emmanuel Pascal Bruno <[EMAIL PROTECTED]>wrote:

> Oh ok, I knew it was something like that.  I have tried many different
> settings on my router.  I'll dig into it some more.
>
> Thanks
>
>
>
> On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>
>> Emmanuel Pascal Bruno wrote:
>> > I have a DID from IPKall.com which is forwarded to my asterisk box.
>> > Then this extension should call my ip phone using Dial application.
>> > Everything works fine, except when I pickup the phone, I can talk, the
>> > other party can hear me, but I cannot hear anything the person says on
>> > the ip phone.
>> > Then after a couple of seconds, the call hangs up.  I don't know why.
>> >
>> > Here is the message I get:
>> >
>> >  SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
>> > -- Native bridging SIP/XX.XX.XXX.XX-09400918 and
>> SIP/ipphone-09401f10
>> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
>> > retries exceeded on transmission
>> > [EMAIL PROTECTED] for seqno 102 (Critical
>> > Response) -- See doc/sip-retransmit.txt.
>> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging
>> > up call [EMAIL PROTECTED] - no reply to
>> > our critical packet (see doc/sip-retransmit.txt).
>> >   == Spawn extension (ipkall, ipphone, 1) exited non-zero on
>> > 'SIP/XX.XX.XXX.XX-09400918'
>> >
>> > I am running asterisk 1.6 on CentOS
>> >
>> > Please help me fix this
>>
>> You likely have firewall issues since it appears that you are not
>> receiving a response from the other end.  Make sure you have *both* your
>> SIP and RTP ports forwarded to your Asterisk box.
>>
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>>
>
>
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Re: [asterisk-users] Call problems

[Emmanuel Pascal Bruno]

Oh ok, I knew it was something like that.  I have tried many different
settings on my router.  I'll dig into it some more.

Thanks


On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:

> Emmanuel Pascal Bruno wrote:
> > I have a DID from IPKall.com which is forwarded to my asterisk box.
> > Then this extension should call my ip phone using Dial application.
> > Everything works fine, except when I pickup the phone, I can talk, the
> > other party can hear me, but I cannot hear anything the person says on
> > the ip phone.
> > Then after a couple of seconds, the call hangs up.  I don't know why.
> >
> > Here is the message I get:
> >
> >  SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
> > -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10
> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
> > retries exceeded on transmission
> > [EMAIL PROTECTED] for seqno 102 (Critical
> > Response) -- See doc/sip-retransmit.txt.
> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging
> > up call [EMAIL PROTECTED] - no reply to
> > our critical packet (see doc/sip-retransmit.txt).
> >   == Spawn extension (ipkall, ipphone, 1) exited non-zero on
> > 'SIP/XX.XX.XXX.XX-09400918'
> >
> > I am running asterisk 1.6 on CentOS
> >
> > Please help me fix this
>
> You likely have firewall issues since it appears that you are not
> receiving a response from the other end.  Make sure you have *both* your
> SIP and RTP ports forwarded to your Asterisk box.
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Call problems

[Emmanuel Pascal Bruno]

I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then after a couple of seconds, the call hangs up.  I don't know why.

Here is the message I get:

 SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
-- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up
call [EMAIL PROTECTED] - no reply to our
critical packet (see doc/sip-retransmit.txt).
  == Spawn extension (ipkall, ipphone, 1) exited non-zero on
'SIP/XX.XX.XXX.XX-09400918'

I am running asterisk 1.6 on CentOS

Please help me fix this
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[asterisk-users] Call problems

[Emmanuel Pascal Bruno]

I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then after a couple of seconds, the call hangs up.  I don't know why.

Here is the message I get:

 SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
-- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED]<[EMAIL PROTECTED]>for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up
call [EMAIL PROTECTED]<[EMAIL PROTECTED]>-
no reply to our critical packet (see doc/sip-retransmit.txt).
  == Spawn extension (ipkall, ipphone, 1) exited non-zero on
'SIP/XX.XX.XXX.XX-09400918'

I am running asterisk 1.6 on CentOS

Please help me fix this
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Re: [asterisk-users] Digium training course

[Pascal Bruno]

That is good you have all those years of experiences and you might know more
than the instructor.  But I dont see the connection, or the point you are
trying to make.  The question is that there is a space to apply a coupon
code, and I was wondering how and where one could get one.  I don't recall
asking for free training, so I don't see why you are saying for that matter
you think people with experience should get the dCAP.  It doesn't make any
sense to me.



On Wed, Sep 17, 2008 at 10:42 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:

> On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno <[EMAIL PROTECTED]> wrote:
> > Anybody knows how to get a Coupon Code for the discount on the Asterisk
> > training classes???  I am interested on taking that upcoming Asterisk
> > Advance course, and 3K is kinda steep and considering I am still a
> college
> > student paying this training out of my pocket, every bit helps.
>
> Sorry to thread jack.
>
> For that matter, I think old timers like myself should automatically
> get a dCAP.
>
> Six or seven years of Asterisk extensive experience should grandfather
> the dCAP and maybe even the training.
>
> I am sure I have a few tricks up my sleeve that the instructors don't know.
>
> If memory serves me correctly, there was talk about this very issue
> when the training and dCAP track came out.  I will google it later.
>
> Thanks
> Steve Totaro
> 1.888.777.1888
>
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[asterisk-users] Digium training course

[Pascal Bruno]

Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes???  I am interested on taking that upcoming Asterisk
Advance course, and 3K is kinda steep and considering I am still a college
student paying this training out of my pocket, every bit helps.
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Re: [asterisk-users] Asterisk and Fedora 9

[Pascal Bruno]

Thanks Jonn!!!



On Fri, Sep 12, 2008 at 2:02 PM, Jonn R Taylor <[EMAIL PROTECTED]>wrote:

>  http://www.taylortelephone.com/asterisk/
>
>
>
> There are install scripts for Centos 5 Asterisk 1.4. They should work just
> fine on FC9. If you have a problem just email me.
>
>
>
> Jonn
>
>
>  --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Pascal Bruno
> *Sent:* Friday, September 12, 2008 9:14 AM
> *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk and Fedora 9
>
>
>
> Ok very good,  how about for the asterisk addonds and sounds?  Can you
> provide me the commands to get, build and install for the 1.4.21 version?
> Thanks a lot guys.
>
>  On Fri, Sep 12, 2008 at 6:07 AM, MFH <[EMAIL PROTECTED]> wrote:
>
> The best way I can think of is:
>
>  >wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz
>  >tar -zxvf asterisk-1.4.21.2.tar.gz
>  >cd asterisk-1.4.21.2
>  >./configure
>  >make menuselect (You don't have to select anything)
>  >make
>  >make install
>  >make samples
>
>
> Pascal Bruno wrote:
> > I am about to install Asterisk on a Fedora 9 box, but i see with yum,
> > they only have Asterisk 1.6 beta in the package repos which I didn't
> > really want to install until they have a stable release.  Does anybody
> > know or have a good and easy way to install Asterisk 1.4 on fedora 9?
> > Thank you.
>
> > 
>
> >
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Re: [asterisk-users] Asterisk and Fedora 9

[Pascal Bruno]

Ok very good,  how about for the asterisk addonds and sounds?  Can you
provide me the commands to get, build and install for the 1.4.21 version?
Thanks a lot guys.


On Fri, Sep 12, 2008 at 6:07 AM, MFH <[EMAIL PROTECTED]> wrote:

> The best way I can think of is:
>
>  >wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz
>  >tar -zxvf asterisk-1.4.21.2.tar.gz
>  >cd asterisk-1.4.21.2
>  >./configure
>  >make menuselect (You don't have to select anything)
>  >make
>  >make install
>  >make samples
>
> Pascal Bruno wrote:
> > I am about to install Asterisk on a Fedora 9 box, but i see with yum,
> > they only have Asterisk 1.6 beta in the package repos which I didn't
> > really want to install until they have a stable release.  Does anybody
> > know or have a good and easy way to install Asterisk 1.4 on fedora 9?
> > Thank you.
> > 
> >
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