[asterisk-users] How to execute Asterisk Functions in PHPAGI

[Zhang Shukun]

Hi, all
I can use $agi->exec() to excute applications in Asterisk.
such as $agi->exec("Set",abc=1)

But how could i excute Asterisk Functions use agi functions?  for example:

I want to excute SHARED function in PHPAGI. i use
$agi->exec("SHARED",callernum) to do it.

but failed with info:

AGI Rx << EXEC SHARED callernum
-- AGI Script Executing Application: (SHARED) Options: (callernum)
[Oct 19 00:06:13] WARNING[2638]: res_agi.c:1757 handle_exec: Could not
find application (SHARED)

Could you help me?
-- 
Appreciate your kindly advise and help.
Thanks & Regards
Sucan

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Re: [asterisk-users] Quintum Tenor AX and Echo

[Antonio Berrios]

  On 10/11/2010 09:22 AM, Antonio Berrios wrote:
>Let's try this again.
>
> I have a Quintum AX Tenor gateway sending calls to Asterisk from BT
> analogue lines connected to FXO.
> The agents hear an echo on their side but incoming callers hear the
> conversation fine. I can't seem to find the problem. Anyone seen this
> issue before?
>
>
>
>
> 
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Turning Rx Gain in the "CAS Signaling Group" for the lines from its 
default of 6dB to 0dB in the Quintum looks to have semi-solved the issue 
of echo on the Asterisk agent's lines.



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Re: [asterisk-users] clustering

[Rizwan Hisham]

anymore ideas anyone please?

On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  wrote:

>  use camailio for SIP SLB
> sip load balancer
>
>
> - Original Message -
> *From:* Rizwan Hisham 
> *To:* Asterisk Users Mailing List - Non-Commercial 
> Discussion
> *Sent:* Thursday, October 14, 2010 5:01 PM
> *Subject:* [asterisk-users] clustering
>
> Hi all,
> I am planning to do clustering for my company's asterisk servers. I dont
> know much about it, just read some articles on the internet and learned how
> to use DUNDi and some basic information about clustering.
> What I need to know is:
> 1. can i register end user with multiple asterisk servers at a time?
> 2. If not, Can I re-route registeration requests to different servers using
> 1 asterisk server as a gateway and multiple clustered asterisk servers
> behind it?
>
> cheers
> Thanks in advance
>
> --
> Best Regards
> Rizwan Qureshi
>
>
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> http://www.eset.com
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Re: [asterisk-users] clustering

[Gareth Blades]

Use camailio or opensips as the registrar server so it accepts the sip 
registrations. You can have copies running on a couple of boxes using 
either a shared databases or a database on each server configured in 
master-master replication mode. Opensips can be configured to use the 
same database table that asterisk uses for authentication. Then you can 
use the load balancer module to send the call to whichever asterisk box 
has the most free lines.
Normally you try and use opensips for most things such as call routing 
and registrations and leave asterisk to do the application type stuff 
such as conference calls and voicemail.

Rizwan Hisham wrote:
> anymore ideas anyone please?
> 
> On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  > wrote:
> 
> use camailio for SIP SLB
> sip load balancer
>  
> 
> - Original Message -
> *From:* Rizwan Hisham 
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> 
> *Sent:* Thursday, October 14, 2010 5:01 PM
> *Subject:* [asterisk-users] clustering
> 
> Hi all,
> I am planning to do clustering for my company's asterisk
> servers. I dont know much about it, just read some articles on
> the internet and learned how to use DUNDi and some basic
> information about clustering.
> What I need to know is:
> 1. can i register end user with multiple asterisk servers at a time?
> 2. If not, Can I re-route registeration requests to different
> servers using 1 asterisk server as a gateway and multiple
> clustered asterisk servers behind it?
> 
> cheers
> Thanks in advance
> 
> -- 
> Best Regards
> Rizwan Qureshi
> 
> 
> 
> 
> 
> -- 
> _
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> http://www.api-digital.com --
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> Thurs:
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> 
> 
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> Le message a iti virifii par ESET NOD32 Antivirus.
> 
> http://www.eset.com
> 
> 
> 
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> 
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> 
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> 
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> 
> 
> 
> 
> -- 
> Best Regards
> Rizwan Qureshi
> 
> 


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Re: [asterisk-users] DMTF Mode

[Dan Journo]

> I recommend reviewing [1] and look for possible regressions.  There
have been some update to RFC2833 over the last few months.

In 1.4.36, its a regression issue. I switched to 1.4.35 and no problems since.
Can anyone else confirm this for me?

Problems are either no dtmf tones being processed by the provider. Or dtmf 
tones getting "stuck" and playing for 3 to 5 seconds before timing out.

I've looked at the tcpdump logs for both, and don't know enough to see where 
the problem could be.

Thanks
Dan

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[asterisk-users] SIP DNS SRV

[Jonas Kellens]

Hello list.

When using SIP DNS SRV to define a production Asterisk server with high 
priority and a backup Asterisk server with a lower priority on this 
DNS-server, will this work as follow :


- production server is reachable, so registration of the IP-phone goes 
to this server
- production server is unreachable, so registration goes to the backup 
Asterisk server
- production server is reachable again, so registration goes back to the 
production server


??

Do I need a low REGISTER timeout value for this to work ? Something like 
60seconds, so the IP-phones register every 60 seconds...




Kind regards,
Jonas.
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Re: [asterisk-users] clustering

[Andrew Latham]

Please read about using DNS and SRV records in the infrastructure.
You can use then to get 95% of the way.


~
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lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
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On Mon, Oct 18, 2010 at 7:29 AM, Rizwan Hisham  wrote:
> anymore ideas anyone please?
>
> On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  wrote:
>>
>> use camailio for SIP SLB
>> sip load balancer
>>
>>
>> - Original Message -
>> From: Rizwan Hisham
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Sent: Thursday, October 14, 2010 5:01 PM
>> Subject: [asterisk-users] clustering
>> Hi all,
>> I am planning to do clustering for my company's asterisk servers. I dont
>> know much about it, just read some articles on the internet and learned how
>> to use DUNDi and some basic information about clustering.
>> What I need to know is:
>> 1. can i register end user with multiple asterisk servers at a time?
>> 2. If not, Can I re-route registeration requests to different servers
>> using 1 asterisk server as a gateway and multiple clustered asterisk servers
>> behind it?
>>
>> cheers
>> Thanks in advance
>>
>> --
>> Best Regards
>> Rizwan Qureshi
>>
>>
>> 
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>    http://www.asterisk.org/hello
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>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> __ Information provenant d'ESET NOD32 Antivirus, version de la
>> base des signatures de virus 5531 (20101014) __
>>
>> Le message a iti virifii par ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>> __ Information provenant d'ESET NOD32 Antivirus, version de la
>> base des signatures de virus 5531 (20101014) __
>>
>> Le message a été vérifié par ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>> --
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>               http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Best Regards
> Rizwan Qureshi
>
>
>
> --
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>

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Re: [asterisk-users] SIP DNS SRV

[Andrew Latham]

You should plan for 2+ productions systems and use DNS round robin to
load balance.  The SRV records define the service but not the path.


~
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lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Mon, Oct 18, 2010 at 7:59 AM, Jonas Kellens  wrote:
> Hello list.
>
> When using SIP DNS SRV to define a production Asterisk server with high
> priority and a backup Asterisk server with a lower priority on this
> DNS-server, will this work as follow :
>
> - production server is reachable, so registration of the IP-phone goes to
> this server
> - production server is unreachable, so registration goes to the backup
> Asterisk server
> - production server is reachable again, so registration goes back to the
> production server
>
> ??
>
> Do I need a low REGISTER timeout value for this to work ? Something like
> 60seconds, so the IP-phones register every 60 seconds...
>
>
>
> Kind regards,
> Jonas.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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[asterisk-users] Asterisk to switch on electric heaters remotely?

[Gilles]

Hello

I'm sure someone has already tried this: I use a couple of electric
heaters to heat my office.

I'd like to somehow connect them to Asterisk so that I could switch
them on remotely by either calling the IVR or sending an e-mail to the
Asterisk host, so that the room is warm when I get to the office :-)

Any information appreciated.

Thank you.


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Re: [asterisk-users] SIP DNS SRV

[Gareth Blades]

Yes that is the way it is supposed to work. You do have to rely on the 
sip devices you are using fully supporting SRV records though.

Jonas Kellens wrote:
>Hello list.
> 
> When using SIP DNS SRV to define a production Asterisk server with high 
> priority and a backup Asterisk server with a lower priority on this 
> DNS-server, will this work as follow :
> 
> - production server is reachable, so registration of the IP-phone goes 
> to this server
> - production server is unreachable, so registration goes to the backup 
> Asterisk server
> - production server is reachable again, so registration goes back to the 
> production server
> 
> ??
> 
> Do I need a low REGISTER timeout value for this to work ? Something like 
> 60seconds, so the IP-phones register every 60 seconds...
> 
> 
> 
> Kind regards,
> Jonas.
> 


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Gareth Blades]

Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only 
control 750W so you will probably need to get it to control a more 
powerfull relay as a heater is going to take a lot of current.
It can be controlled by a virtual serial port so you just program the 
extension to make a system() call to a simple script which sends a 
string of characters to the serial port.

That device is quite expensive. You could probably find something much 
cheaper on ebay.


Gilles wrote:
> Hello
> 
> I'm sure someone has already tried this: I use a couple of electric
> heaters to heat my office.
> 
> I'd like to somehow connect them to Asterisk so that I could switch
> them on remotely by either calling the IVR or sending an e-mail to the
> Asterisk host, so that the room is warm when I get to the office :-)
> 
> Any information appreciated.
> 
> Thank you.
> 
> 


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Roberto Piola]

we're using a Damocles Mini
(http://www.hw-group.com/products/damocles/damocles_mini_en.html). of
course, the damocles will have to drive a high-power relay.

the damocles can be driven via snmp, so you'll have to simply call the
snmpset unix standard utility

On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades
 wrote:
> Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only
> control 750W so you will probably need to get it to control a more
> powerfull relay as a heater is going to take a lot of current.
> It can be controlled by a virtual serial port so you just program the
> extension to make a system() call to a simple script which sends a
> string of characters to the serial port.
>
> That device is quite expensive. You could probably find something much
> cheaper on ebay.
>
>
> Gilles wrote:
>> Hello
>>
>> I'm sure someone has already tried this: I use a couple of electric
>> heaters to heat my office.
>>
>> I'd like to somehow connect them to Asterisk so that I could switch
>> them on remotely by either calling the IVR or sending an e-mail to the
>> Asterisk host, so that the room is warm when I get to the office :-)
>>
>> Any information appreciated.
>>
>> Thank you.
>>
>>
>
>
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[asterisk-users] a2billing

[Baha @ SH]

Not sure if a2billing can be shared here, but ill give a shot

If the credit < min_credit the IVR play: sorry you have 0 credit and hangup,
I want it to FW me to the IVR to add voucher, please let me know: here is
log:

[18/10/2010 07:01:12]:[file:a2billing.php -
line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
[18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE
: standard]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
[18/10/2010 07:01:12]:[file:a2billing.php -
line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
activated, inuse, simultaccess, typepaid, creditlimit, language,
removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
tariff=cc_tariffgroup.id WHERE username='9971524976']
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
language=en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
prepaid-zero-balance (cardnumber:9971524976)]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
callingcard_ivr_authenticate]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
(cia_res:-2)]]
[18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]





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Re: [asterisk-users] SIP DNS SRV

[Jonas Kellens]

Hello,

I know YeaLink for example supports this...


Can you tell me for sure that when the production Asterisk server 
becomes reachable again, the registration will go back to the production 
server ??



Jonas.



On 10/18/2010 01:11 PM, Gareth Blades wrote:

Yes that is the way it is supposed to work. You do have to rely on the
sip devices you are using fully supporting SRV records though.


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Re: [asterisk-users] fraud advice

[SIP]

  On 10/14/10 9:10 PM, Jeff LaCoursiere wrote:
> Hi,
>
> Embarrassed as I am to write this, I am hoping for some advice.  One of
> our very first PBX installs, now six years old, was "taken advantage of"
> over the past few weeks.  A victim of sipvicious, I assume, that managed
> to guess one of the SIP passwords.  4000 calls to various middle eastern
> destinations have been placed, which ended up being sent over our
> customer's PSTN trunk, and of course there was no warning until the bill
> came today.  Unfortunately the bill only covered the first few days of
> this fiasco, and was only $700.  I am afraid the one that is on the way
> will be tens of thousands.  ONE CALL on the bill that just arrived was
> $200 (80 minutes to Sierra Leone).
>
> I'm sure this started out as a single scan.  It must have been posted,
> because I have at least ten IP addresses now that were placing calls via
> the same peer.  They are from all over the world.
>
> So what is the accepted procedure?  I'm in the US Virgin Islands, so do I
> go to the FBI?  Police?  Is their some telecom fraud body to report such
> things to?  Does any one ever get any relief from such events?
>
> I'm basically sick to my stomach right now.
>
> j
>
We were hit several times in our early days with PRS fraud that ended up 
costing us DEARLY. We contacted the FBI, but they were completely 
unhelpful. The origin of the caller was Egypt (using a network in Egypt 
that has long been a front for criminal activity, so the networking 
people on that end were less than useless), and the Egyptian cyber fraud 
division is two guys with a yahoo email address. The FBI contacted them, 
but they were neither equipped nor entirely willing to be of any real 
help in tracking down the perpetrator. It doesn't hurt to contact the 
FBI, though. They may already have an open investigation into the 
individual or group responsible and need the information for their case. 
But do not expect them to be able to do much.

Eventually, some of our debt was quashed by the provider who had 
violated their own policies in charging us for unlisted premium rate 
services, but it changed the entire way we do business.

Unfortunately, it's now MUCH more difficult to pay us money than it used 
to be, and that's turned a lot of customers off, but we've had no 
problems with PRS fraud since.


N.

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Re: [asterisk-users] SIP DNS SRV

[Gareth Blades]

Yes if you set the production server to a higher priority then the 
backup then the RFC states that the client should periodically check and 
switch back to the primary server is it becomes reachable.


Jonas Kellens wrote:
>Hello,
> 
> I know YeaLink for example supports this...
> 
> 
> Can you tell me for sure that when the production Asterisk server 
> becomes reachable again, the registration will go back to the production 
> server ??
> 
> 
> Jonas.
> 
> 
> 
> On 10/18/2010 01:11 PM, Gareth Blades wrote:
>> Yes that is the way it is supposed to work. You do have to rely on the 
>> sip devices you are using fully supporting SRV records though.
> 


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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Jigar Joshi]

Thanks Gopalakrishnan ,Sebastian,Josef,

Actually My Requirement is :

I need a international number all network should be able to connect to it.
After ringing a ring call should be picked up. and should ask for a code.
code should come from mysql or any other DB
depending upon the code it should route the call to an application where it
converts SIP to my protocol and vica versa.

I am confused about the implementation. design is clear ,

Your help would be really helpful.




On Sun, Oct 17, 2010 at 12:08 AM, Sebastian  wrote:

>
>
> On 10/14/2010 03:09 PM, Gopalakrishnan A.N wrote:
> > Hi Joshi,
> >
> > To connect with PSTN line you need FXO / FXS card. FXO is used to
> > connect CO line and FXS is used to connect internal station line. With
> > help of FXO you can connect the outside world and with help of FXS you
> > can connect normal analog phones. Inspite of normal analog phones you
> > can connect SIP phones (soft phones) also.
>
> Actually to connect PSTN lines (regular telephone lines coming from your
> telecom provider) to Asterisk you only need FXO cards. Or ATA's
> (analogue telephone adapters) - specially if your Asterisk box doesn't
> have PCI or PCI-e slots. The FXO cards can be PCI or PCI-express (or
> other flavours for more complex setups) with 1-8 (or even more)
> connectors per card for the same number of telephone lines to be
> connected to Asterisk. Digium, Sangoma, OpenVox and others make them.
>
> You only need FXS cards if you have analogue phones which you want to
> re-use. Otherwise there are many manufacturers of SIP hardware phones -
> looking perfectly similar to regular business desk phones - so there
> isn't much point IMO to buy FSX cards unless you already have the
> analogue phones and need to re-use them.
>
> Sebastian
>
>
> Some vendors are there for
> > these PSTN cards like Digium, Sangoma, Openvox.
> >
> > Good luck:)
> >
> >
> >
> > On Thu, Oct 14, 2010 at 7:05 PM, Jigar Joshi  > > wrote:
> >
> > Hello community,
> >
> > I have successfully set up asterisk free PBX server and I am also
> > able to connect to it by softphone.
> >
> > Now as next step I want to extend this to PSTN ,
> >
> > My Required scenario:
> >
> > I need a number which will connect outside PSTN world to my PBX and
> > by applying extension particular softphone or connected normal phone
> > should get connected.
> >
> > Which hardware I need for it.
> > Also please explain a bit of dial plans.
> >
> > Thanks
> >
> >
> >
> > --
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> >
> >
> >
> >
> > --
> > Thank you  with regards,
> > Gopalakrishnan A.N,
> >
> >
>
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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Gilles]

On Mon, 18 Oct 2010 17:36:30 +0530, Jigar Joshi 
wrote:
>I need a international number all network should be able to connect to it.
>After ringing a ring call should be picked up. and should ask for a code.
>code should come from mysql or any other DB
>depending upon the code it should route the call to an application where it
>converts SIP to my protocol and vica versa.

www.google.com/search?q=voip+provider


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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Jigar Joshi]

Gillies, Can't I configure Asterisk for the same on my live IP system. ?

I am very newbie to all this .

On Mon, Oct 18, 2010 at 5:48 PM, Gilles  wrote:

> On Mon, 18 Oct 2010 17:36:30 +0530, Jigar Joshi 
> wrote:
> >I need a international number all network should be able to connect to it.
> >After ringing a ring call should be picked up. and should ask for a code.
> >code should come from mysql or any other DB
> >depending upon the code it should route the call to an application where
> it
> >converts SIP to my protocol and vica versa.
>
> www.google.com/search?q=voip+provider
>
>
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Re: [asterisk-users] SIP DNS SRV

[Philipp von Klitzing]

Hi!

> Do I need a low REGISTER timeout value for this to work ? Something
> like 60seconds, so the IP-phones register every 60 seconds... 

This can also be done without SRV records if you have phones that provide 
a fallback SIP registrar entry - snom for example does, and many other 
vendors do to.

Philipp


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Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

[Karsten Wemheuer]

Am Samstag, den 16.10.2010, 14:00 -0500 schrieb Shaun Ruffell:
> On 10/16/10 12:47 PM, Karsten Wemheuer wrote:
> > Hi,
> >
> > Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell:
> >> On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
> >>
> >>> I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
> >>> dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
> >>> older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
> >>> card. The system starts without any errors.
> >>>
> >>> I discovered a severe issue. The kernel panics on a very small load. The
> >>> first call normally gets through. If I start the second or third call
> >>> and sometimes when I terminate the first call, the system panics (Oops
> >>> text on console).
> >>>
> >>> After solving some difficulties (the relevant part of the Oops text
> >>> scrolls out of the monitor, no serial interface), I get the text via
> >>> netconsole. It seems to me, that the panic occurred in oslec (function
> >>> "oslec_update"). But maybe I am wrong with this. In the oslec code there
> >>> is a patch to enable MMX. After switching this off, the problem
> >>> disappeared. AFAIK the cpu supports mmx.
> >>>
> >>> Where should I address this issue to? Is it a known issue?
> >>>
> >>
> >> Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h?
> >
> > No, I don't think so. (This is from memory, I currently have no access
> > to the test system). But there was a patch (I think from debian
> > packages, original from bug tracker (dahdi_mmx_auto.diff from
> > http://bugs.digium.com/view.php?id=13500)) which enables mmx at least
> > for the echo canceler oslec (I think). Disabling this patch let the
> > kernel panic disappear.
> >
> 
> Hmmm...I can't be certain since there are many parts coming from out of 
> the tree here (besides just olsec itself), but looking at 
> https://issues.asterisk.org/file_download.php?file_id=22366&type=bug 
> doesn't *seem* right.  It appears that DAHDI_USE_MMX is exported and 
> therefore the olsec in git://gitorious.org/dahdi-extra/dahdi-extra.git 
> uses the MMX instructions, but -DCONFIG_DAHDI_MMX is only added onto 
> CFLAGS_zaptel_base.o and not CFLAGS_dahdi_base.o.  Therefore, oslec most 
> likely is killing the FPU registers since it believes that dahdi-base.c 
> is taking care of saving and restoring them by hand.
> 
> I would recommend changing CFLAGS_zaptel_base.o to CFLAGS_dahdi_base.o, 
> or hand edit include/dahdi/dahdi_config.h to make sure CONFIG_DAHDI_MMX 
> is defined and see if you still get the crash.

I changed CFLAGS_zaptel_base.o to CFLAGS_dahdi_base.o as You recommended
and it seems to work now. No crash anymore.

Thanks,

Karsten



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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Gilles]

On Mon, 18 Oct 2010 17:54:27 +0530, Jigar Joshi 
wrote:
>Gillies, Can't I configure Asterisk for the same on my live IP system. ?

I don't understand what you mean.

To let Asterisk get calls from the phone network (POTS" a.k.a. PSTN),
you either need a phone line + PCI card or ATA to connect the PC to
the phone line, or you get a subscription with a VoIP provider which
will provide you with a phone number and ring your Asterisk box
through the Net.


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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Jigar Joshi]

Thanks Gilles You got it exactly right,

Once I receive call,

I will let user enter a code.

All the incoming calls will resolve to one extension.

and there one app will be running that will parse the code and will process
it there.

What technically i need is :

1. An international number , [That you told ,we 'll get it from VIOP
providers] ,I will work on it

2 Configuration that will stream all call ,[all incoming calls with any
extension to a application running on machine, with code entered by user in
each packet]

3.I will process those packet and send it back on the protocol which
asterisk speaks.

I hope it makes sense.

by the way Thanks a lot for your quick reply.


On Mon, Oct 18, 2010 at 6:53 PM, Gilles  wrote:

> On Mon, 18 Oct 2010 17:54:27 +0530, Jigar Joshi 
> wrote:
> >Gillies, Can't I configure Asterisk for the same on my live IP system. ?
>
> I don't understand what you mean.
>
> To let Asterisk get calls from the phone network (POTS" a.k.a. PSTN),
> you either need a phone line + PCI card or ATA to connect the PC to
> the phone line, or you get a subscription with a VoIP provider which
> will provide you with a phone number and ring your Asterisk box
> through the Net.
>
>
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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Gilles]

On Mon, 18 Oct 2010 19:12:48 +0530, Jigar Joshi 
wrote:
>1. An international number , [That you told ,we 'll get it from VIOP
>providers] ,I will work on it

VoIP provider.

>2 Configuration that will stream all call ,[all incoming calls with any
>extension to a application running on machine, with code entered by user in
>each packet]

www.google.com/search?q=asterisk+ivr


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Steve Edwards]

On Mon, 18 Oct 2010, Gilles wrote:

> I'd like to somehow connect them to Asterisk so that I could switch
> them on remotely by either calling the IVR or sending an e-mail to the
> Asterisk host, so that the room is warm when I get to the office :-)

X10.

(X10's web site, x10.com sucks.)

http://www.smarthome.com/2414U/PowerLinc-INSTEON-Controller-USB-Based-Home-Automation-Device/p.aspx
http://www.smarthome.com/2477SA1/INSTEON-220V-240V-30-AMP-Load-Controller-Normally-Open-Relay-Dual-Band/p.aspx

Not a big fan of the idea though. What if someone tossed their coat over 
the heater the previous day? What if you get hit by a truck on your way 
in and the heater runs continuously until...?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Marco Signorini]

Hi
Did you looked at Arduino + Ethernet Shield?
Is something you can program in C or C++ to receive a simple TCP and/or
HTTP packet and turn on an external relay.
>From the dialplan you can run an http query through curl and/or an
external AGI command.

Best regards,
Marco Signorini.

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http://www.ingegnitech.com


Roberto Piola wrote:
> we're using a Damocles Mini
> (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of
> course, the damocles will have to drive a high-power relay.
>
> the damocles can be driven via snmp, so you'll have to simply call the
> snmpset unix standard utility
>
> On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades
>  wrote:
>   
>> Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only
>> control 750W so you will probably need to get it to control a more
>> powerfull relay as a heater is going to take a lot of current.
>> It can be controlled by a virtual serial port so you just program the
>> extension to make a system() call to a simple script which sends a
>> string of characters to the serial port.
>>
>> That device is quite expensive. You could probably find something much
>> cheaper on ebay.
>>
>>
>> Gilles wrote:
>> 
>>> Hello
>>>
>>> I'm sure someone has already tried this: I use a couple of electric
>>> heaters to heat my office.
>>>
>>> I'd like to somehow connect them to Asterisk so that I could switch
>>> them on remotely by either calling the IVR or sending an e-mail to the
>>> Asterisk host, so that the room is warm when I get to the office :-)
>>>
>>> Any information appreciated.
>>>
>>> Thank you.
>>>
>>>
>>>   
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>>
>> 
>
>
>
>   


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Re: [asterisk-users] clustering

[Rizwan Hisham]

Unfortunately we are too late to switch to Kamailio. I mean we have
developed our pbx with call features and routing on asterisk only. If we
switch to some other software that means we will have to redo a lot of
development again. I was thinking of using DUNDi and distributing the
registrations on different servers.

I just dont get one point. lets say if i have 2 users registered on
different asterisk servers and one of the server fails (dundi doea not get
anything in return from lookup). But I can still get the contact information
for the user who was registered on the failed server from db (realtime peer)
for incoming calls. But what happens when that user tries to make a outgoing
call? How do I redirect the call to the server which is still working?

On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades
wrote:

> Use camailio or opensips as the registrar server so it accepts the sip
> registrations. You can have copies running on a couple of boxes using
> either a shared databases or a database on each server configured in
> master-master replication mode. Opensips can be configured to use the
> same database table that asterisk uses for authentication. Then you can
> use the load balancer module to send the call to whichever asterisk box
> has the most free lines.
> Normally you try and use opensips for most things such as call routing
> and registrations and leave asterisk to do the application type stuff
> such as conference calls and voicemail.
>
> Rizwan Hisham wrote:
> > anymore ideas anyone please?
> >
> > On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  > > wrote:
> >
> > use camailio for SIP SLB
> > sip load balancer
> >
> >
> > - Original Message -
> > *From:* Rizwan Hisham 
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > *Sent:* Thursday, October 14, 2010 5:01 PM
> > *Subject:* [asterisk-users] clustering
> >
> > Hi all,
> > I am planning to do clustering for my company's asterisk
> > servers. I dont know much about it, just read some articles on
> > the internet and learned how to use DUNDi and some basic
> > information about clustering.
> > What I need to know is:
> > 1. can i register end user with multiple asterisk servers at a
> time?
> > 2. If not, Can I re-route registeration requests to different
> > servers using 1 asterisk server as a gateway and multiple
> > clustered asterisk servers behind it?
> >
> > cheers
> > Thanks in advance
> >
> > --
> > Best Regards
> > Rizwan Qureshi
> >
> >
> >
> 
> >
> > --
> >
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> > Thurs:
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > __ Information provenant d'ESET NOD32 Antivirus, version
> > de la base des signatures de virus 5531 (20101014) __
> >
> > Le message a iti virifii par ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> >
> >
> > __ Information provenant d'ESET NOD32 Antivirus, version de
> > la base des signatures de virus 5531 (20101014) __
> >
> > Le message a été vérifié par ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> > --
> > _
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> >
> >
> >
> >
> > --
> > Best Regards
> > Rizwan Qureshi
> >
> >
>
>
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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Danny Nicholas]

Mindless technology is only as good as the weakest mind that develops and
uses it...


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Re: [asterisk-users] clustering

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Monday, October 18, 2010 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] clustering

 

Unfortunately we are too late to switch to Kamailio. I mean we have
developed our pbx with call features and routing on asterisk only. If we
switch to some other software that means we will have to redo a lot of
development again. I was thinking of using DUNDi and distributing the
registrations on different servers.

I just dont get one point. lets say if i have 2 users registered on
different asterisk servers and one of the server fails (dundi doea not get
anything in return from lookup). But I can still get the contact information
for the user who was registered on the failed server from db (realtime peer)
for incoming calls. But what happens when that user tries to make a outgoing
call? How do I redirect the call to the server which is still working?

On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades 
wrote:

Use camailio or opensips as the registrar server so it accepts the sip
registrations. You can have copies running on a couple of boxes using
either a shared databases or a database on each server configured in
master-master replication mode. Opensips can be configured to use the
same database table that asterisk uses for authentication. Then you can
use the load balancer module to send the call to whichever asterisk box
has the most free lines.
Normally you try and use opensips for most things such as call routing
and registrations and leave asterisk to do the application type stuff
such as conference calls and voicemail.




>From what I have read, Asterisk and Kamalio can co-exist;  some posters have
used Kamalio to "supplement" the features that Asterisk either doesn't
provide or doesn't provide in as nice a form as the OP desired - can't
really speak beyond this as I am not one of them.

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Re: [asterisk-users] clustering

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Monday, October 18, 2010 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] clustering

 

Unfortunately we are too late to switch to Kamailio. I mean we have
developed our pbx with call features and routing on asterisk only. If we
switch to some other software that means we will have to redo a lot of
development again. I was thinking of using DUNDi and distributing the
registrations on different servers.

I just dont get one point. lets say if i have 2 users registered on
different asterisk servers and one of the server fails (dundi doea not get
anything in return from lookup). But I can still get the contact information
for the user who was registered on the failed server from db (realtime peer)
for incoming calls. But what happens when that user tries to make a outgoing
call? How do I redirect the call to the server which is still working?

 

Sorry for second post, but I have a Polycom 501 registered to 3 servers.  I
hit the line button and if the server I pick is down, I don't get a dial
tone.  Hope this is useful.

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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Darren Wiebe]

  We recently completed a project using products from here:  
http://www.controlbyweb.com/webrelay/  They were easy to setup and can 
be controlled in a variety of fashions included http queries.

Darren Wiebe

On 18/10/2010 8:34 AM, Marco Signorini wrote:
> Hi
> Did you looked at Arduino + Ethernet Shield?
> Is something you can program in C or C++ to receive a simple TCP and/or
> HTTP packet and turn on an external relay.
>  From the dialplan you can run an http query through curl and/or an
> external AGI command.
>
> Best regards,
> Marco Signorini.
>
> --
> Marco Signorini
> http://www.ethermania.com
> http://www.ingegnitech.com
>
>
> Roberto Piola wrote:
>> we're using a Damocles Mini
>> (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of
>> course, the damocles will have to drive a high-power relay.
>>
>> the damocles can be driven via snmp, so you'll have to simply call the
>> snmpset unix standard utility
>>
>> On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades
>>   wrote:
>>
>>> Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only
>>> control 750W so you will probably need to get it to control a more
>>> powerfull relay as a heater is going to take a lot of current.
>>> It can be controlled by a virtual serial port so you just program the
>>> extension to make a system() call to a simple script which sends a
>>> string of characters to the serial port.
>>>
>>> That device is quite expensive. You could probably find something much
>>> cheaper on ebay.
>>>
>>>
>>> Gilles wrote:
>>>
 Hello

 I'm sure someone has already tried this: I use a couple of electric
 heaters to heat my office.

 I'd like to somehow connect them to Asterisk so that I could switch
 them on remotely by either calling the IVR or sending an e-mail to the
 Asterisk host, so that the room is warm when I get to the office :-)

 Any information appreciated.

 Thank you.



>>> --
>>> _
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>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>
>>
>>
>


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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Jigar Joshi]

@Gilles here are my requirement.
can you please help me .


On Mon, Oct 18, 2010 at 7:12 PM, Jigar Joshi  wrote:

> Thanks Gilles You got it exactly right,
>
> Once I receive call,
>
> I will let user enter a code.
>
> All the incoming calls will resolve to one extension.
>
> and there one app will be running that will parse the code and will process
> it there.
>
> What technically i need is :
>
> 1. An international number , [That you told ,we 'll get it from VIOP
> providers] ,I will work on it
>
> 2 Configuration that will stream all call ,[all incoming calls with any
> extension to a application running on machine, with code entered by user in
> each packet]
>
> 3.I will process those packet and send it back on the protocol which
> asterisk speaks.
>
> I hope it makes sense.
>
> by the way Thanks a lot for your quick reply.
>
>
> On Mon, Oct 18, 2010 at 6:53 PM, Gilles  wrote:
>
>> On Mon, 18 Oct 2010 17:54:27 +0530, Jigar Joshi 
>> wrote:
>> >Gillies, Can't I configure Asterisk for the same on my live IP system. ?
>>
>> I don't understand what you mean.
>>
>> To let Asterisk get calls from the phone network (POTS" a.k.a. PSTN),
>> you either need a phone line + PCI card or ATA to connect the PC to
>> the phone line, or you get a subscription with a VoIP provider which
>> will provide you with a phone number and ring your Asterisk box
>> through the Net.
>>
>>
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>>   http://www.asterisk.org/hello
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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] clustering

[Zeeshan Zakaria]

How about setting up a high availability cluster using DRBD and Heartbeat?
There is some good info on it on the Internet. In this type of setup you
have two exact same servers running in parallel, and only one has the
required services up. They keep themselves in sync. When the primary one
goes down, the secondary instantly takes over. Active calls are though
dropped, but after that everything is back to normal. There are various
other options regarding which server will stay primary, or how and which
services will be used on which server.

Another option I am exploring is using the same thing but in Proxmox with
DRBD. Somebody told me it could be setup so that even the active calls are
not dropped. I haven't set it up yet, but will try it when get time.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-10-18 10:59 AM, "Danny Nicholas"  wrote:

  --

*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham


Sent: Monday, October 18, 2010 9:43 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] clustering



Unfortunately we are too late to switch to Kamailio. I mean we have
developed our pbx with call features and routing on asterisk only. If we
switch to some other software that means we will have to redo a lot of
development again. I was thinking of using DUNDi and distributing the
registrations on different servers.



I just dont get one point. lets say if i have 2 users registered on
different asterisk servers and...



Sorry for second post, but I have a Polycom 501 registered to 3 servers.  I
hit the line button and if the server I pick is down, I don’t get a dial
tone.  Hope this is useful.

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[asterisk-users] Problems detecting hangup

[Ariel Wainer]

Hello people, this is my first mail to this list. I'm new to asterisk
and trying to set up an IVR. So far my dialplan works nice connecting
with softphones, but I'm having problems to detect hangups on the analog
line.
Here are the details:

-Clone pc with Ubuntu server 10.04 64 bits
-Asterisk 1.6
-Openvox A400P, lspci  -v says:
01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Device b100:0001
Flags: bus master, medium devsel, latency 32, IRQ 21
I/O ports at c800 [size=256]
Memory at ff0ff000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Kernel driver in use: wctdm
Kernel modules: wctdm, wcopenpci, hisax, netjet

And this is the scheme:

 PSTN <-> Analog Panasonic PBX <-> Asterisk


The problem as you already may know, is the busy/hangup tone generated
by the Panasonic PBX, connecting the asterisk boc directly to the pstn
works fine.
The Panasonic model is KX-TA616, as far as I understand, the manual[0]
says that it can detect Polarity reversion, but not generate it.
I recorded the hangup sound (using Record()), converted and opened in
audacity: it's 260ms tone, 250ms
 So I configured:
busydetect=yes
busycount=3
busypattern=260,250

But sill can't detect the hangup. 



The tone can be downloaded here  http://cringer.3kh.net/web/hangup-tone.wav
And the dahdi configuration is here:
http://cringer.3kh.net/web/chan_dahdi.conf
http://cringer.3kh.net/web/dahdi-channels.conf

Let me know if you need any other details on the setup.

Thank you very much in advance.

[0] http://smarthomeuae.com/new3/product.php?productid=218&pcategory=16001 


PS: please excuse my bad english.
-- 
Ariel Wainer
Contenta Mobile


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Lyle Giese]

Gilles wrote:
> Hello
>
> I'm sure someone has already tried this: I use a couple of electric
> heaters to heat my office.
>
> I'd like to somehow connect them to Asterisk so that I could switch
> them on remotely by either calling the IVR or sending an e-mail to the
> Asterisk host, so that the room is warm when I get to the office :-)
>
> Any information appreciated.
>
> Thank you.
>
>
>   
I use a linux box to control the hvac in my home using a QK108 instead
of a conventional thermostat(on a forced air nat gas furnace). I use
1wire probes from www.hobbyboards.com to monitor temperature and
humidity. I wrote custom perl cgi scripts to control this via a webpage.

Up to you from there on how fancy you want to get. I suspose you could
use an IVR system to reset the temperature...

Lyle


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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Steve Edwards]

On Mon, 18 Oct 2010, Jigar Joshi wrote:

> @Gilles here are my requirement.can you please help me .

Are you putting this "out to bid" or are you just too lazy to read ATFOT 
(http://downloads.oreilly.com/books/9780596510480.pdf)?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] SIP DNS SRV

[Jonas Kellens]

To be clear: I need to set the SRV record on my DNS-server and not on my 
Asterisk server ?!


Because the people @ YeaLink keep on talking about "Your SIP server 
needs to support this"...



Jonas.


On 10/18/2010 02:03 PM, Gareth Blades wrote:

Yes if you set the production server to a higher priority then the
backup then the RFC states that the client should periodically check and
switch back to the primary server is it becomes reachable.


Jonas Kellens wrote:
   

Hello,

I know YeaLink for example supports this...


Can you tell me for sure that when the production Asterisk server
becomes reachable again, the registration will go back to the production
server ??


Jonas.


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[asterisk-users] How to check if Agent is logged into a specific Queue using dial-plan?

[bruce bruce]

Hi,

I have this on an Aastra phone:

Button 1:Login English Queue
Button 2:Login French Queue
Button 3:Logout both English and French


I am out of buttons and using only three buttons I want my third button to
be smarter. Currently the third button does a QueueRemoveMember to both
English and French Queue at the same time. I want this button to be smarter
to and to check and see if the Agent is logged into only English to only do
a Remove on English or if the Agent is only logged into French to only log
out French. Same goes for both, if both are logged in to log out both.

Currently I have this for Third Button:

exten => 99,1,Answer
exten =>
99,n,RemoveQueueMember(900|Local/${CALLERID(num)}...@from-internal/n)
exten =>
99,n,RemoveQueueMember(899|Local/${CALLERID(num)}...@from-internal/n)
exten => 99,n,Hangup

900 is English and 800 is Spanish Queue numbers.

P.S. Is there a way to do exten = s rather than exten = 99 as someone
from outside might find out about the 99 extension and try to log into
it? This is for an Aastra phone with XML support.

Thanks
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Re: [asterisk-users] clustering

[Rizwan Hisham]

Hello Zeeshan,
How about doing the mixture of what I want to do with your strategy. I mean,
what if we have 3 asterisk servers with distributed registrations and also
have heartbeat installed monitoring all the servers? will that work?

On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria  wrote:

> How about setting up a high availability cluster using DRBD and Heartbeat?
> There is some good info on it on the Internet. In this type of setup you
> have two exact same servers running in parallel, and only one has the
> required services up. They keep themselves in sync. When the primary one
> goes down, the secondary instantly takes over. Active calls are though
> dropped, but after that everything is back to normal. There are various
> other options regarding which server will stay primary, or how and which
> services will be used on which server.
>
> Another option I am exploring is using the same thing but in Proxmox with
> DRBD. Somebody told me it could be setup so that even the active calls are
> not dropped. I haven't set it up yet, but will try it when get time.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-10-18 10:59 AM, "Danny Nicholas"  wrote:
>
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
>
>
> Sent: Monday, October 18, 2010 9:43 AM
>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] clustering
>
>
>
> Unfortunately we are too late to switch to Kamailio. I mean we have
> developed our pbx with call features and routing on asterisk only. If we
> switch to some other software that means we will have to redo a lot of
> development again. I was thinking of using DUNDi and distributing the
> registrations on different servers.
>
>
>
> I just dont get one point. lets say if i have 2 users registered on
> different asterisk servers and...
>
> 
>
> Sorry for second post, but I have a Polycom 501 registered to 3 servers.  I
> hit the line button and if the server I pick is down, I don’t get a dial
> tone.  Hope this is useful.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
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>   http://www.asterisk.org/hello
>
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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] a2billing

[bruce bruce]

Turn on the voucher feature in System Settings and it will tell the user
right after the PIN authentication or CLID authentication that their balance
is below threshold and they should pay.

-Bruce

On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:

> Not sure if a2billing can be shared here, but ill give a shot
>
> If the credit < min_credit the IVR play: sorry you have 0 credit and
> hangup,
> I want it to FW me to the IVR to add voucher, please let me know: here is
> log:
>
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:76]:[CallerID:]:[CN:]:[MODE
> : standard]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
> SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
> activated, inuse, simultaccess, typepaid, creditlimit, language,
> removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
> expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
> UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
> cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
> cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
> tariff=cc_tariffgroup.id WHERE username='9971524976']
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
> active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
> language=en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
> prepaid-zero-balance (cardnumber:9971524976)]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
> callingcard_ivr_authenticate]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
> (cia_res:-2)]]
> [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]
>
>
>
>
>
> --
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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Daniel Tryba]

On Mon, Oct 18, 2010 at 04:34:34PM +0200, Marco Signorini wrote:
> Did you looked at Arduino + Ethernet Shield?
> Is something you can program in C or C++ to receive a simple TCP and/or
> HTTP packet and turn on an external relay.
> From the dialplan you can run an http query through curl and/or an
> external AGI command.

I second this (as a hobby project). The Arduino starter/exp kit. contains all
necesarry components (a relay and PTC). The relay can switch 5A IIRC but
since I only have to short my thermostat this is not a problem.
Spaceheaters require a heavier relay. Building this is a the combination
of 2 examples (CIRC10 and CIRC11 from
http://ardx.org/src//guide/2/ARDX-EG-OOML-DD.pdf).

-- 

   Daniel Tryba

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Re: [asterisk-users] clustering

[Zeeshan Zakaria]

Hi,

I have worked with only two servers setup, don't know how it would work in
three server setup. You'll need to do an experiment, but know that it won't
work if you have T1 lines. HA and DRBD is good for pure VoIP.

Before the end of this year hopefully I'll be setting up two more redundancy
solutions, and will try some new techniques, and probably try three server
setup too. At that time I plan to post a tutorial on redundancy solution on
my blog, because seems like a lot of people want to know how to do it, yet
guidance is very limited.

Zeeshan A Zakaria

--
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On 2010-10-18 12:49 PM, "Rizwan Hisham"  wrote:

Hello Zeeshan,
How about doing the mixture of what I want to do with your strategy. I mean,
what if we have 3 asterisk servers with distributed registrations and also
have heartbeat installed monitoring all the servers? will that work?



On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria  wrote:
>
> How about setting...
-- 
Best Regards
Rizwan Qureshi



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[asterisk-users] Asterisk 1.8.0 Release Candidate 4 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the fourth release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently
scheduled to become the full release of Asterisk 1.8.0.

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the last release candidate as
reported by the community. A sampling of the changes in this release candidate
include:

  * Additional fixups in chan_gtalk that allow outbound calls to both Google
Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
and stunaddr.
(Closes issue #13971. Patched by dvossel)

  * Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)

  * Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)

  * Resolve issue where faxdetect would only detect the first fax call in
chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)

  * Resolve issue where a channel that is setup and torn down *very* quickly may
not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
 https://reviewboard.asterisk.org/r/740/)

  * Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)

  * Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)

  * Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)

A short list of available features includes:

   * Secure RTP
   * IPv6 Support in the SIP channel driver
   * Connected Party Identification Support
   * Calendaring Integration
   * A new call logging system, Channel Event Logging (CEL)
   * Distributed Device State using Jabber/XMPP PubSub
   * Call Completion Supplementary Services support
   * Advice of Charge support
   * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4

Thank you for your continued support of Asterisk!

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[asterisk-users] CEL Documentation

[Jeremy Betts]

Anybody know where to find some good information on the new CEL in 
asterisk 1.8? I'm very anxious to check out the new logging features but 
can't find anything but the cel.conf.sample file in the source package. 
I'd like to get this setup with ODBC.

Thanks in advance.

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Re: [asterisk-users] CEL Documentation

[Andrew Latham]

I would look at

http://svnview.digium.com/svn/asterisk/trunk/configs/cel.conf.sample?view=markup

and

http://svnview.digium.com/svn/asterisk/trunk/configs/cel_custom.conf.sample?view=markup

they are a good start

~
Andrew "lathama" Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Mon, Oct 18, 2010 at 3:54 PM, Jeremy Betts  wrote:
> Anybody know where to find some good information on the new CEL in
> asterisk 1.8? I'm very anxious to check out the new logging features but
> can't find anything but the cel.conf.sample file in the source package.
> I'd like to get this setup with ODBC.
>
> Thanks in advance.
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>

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[asterisk-users] Recording

[Dan Journo]

Hi,

Does anyone have a professional recording of someone saying "Recording" so I 
can let the operator know that the one-touch recording has started successfully?

Thanks
Dan



[cid:image001.gif@01CB6F04.2EEFF060]



Dan Journo
IT Business Consultant
Kesher Communications Ltd
Tel: 0161 820 8353
Fax: 0161 820 8352
Web: http://www.KesherCommunications.com
Live Chat/Instant Support: Click 
Here

This email and any files transmitted with it are confidential and intended 
solely for the recipient(s). If you are not the named addressee you should not 
disseminate, copy or alter this email. Under no circumstances may this email be 
distributed without written permission from the sender. Warning: Although the 
Company has taken reasonable precautions to ensure no viruses are present in 
this email, the company cannot accept responsibility for any loss or damage 
arising from the use of this email or attachments. All prices exclude VAT 
unless otherwise stated. No responsibility is taken for any recommendations 
made by a sender or by Kesher Communications Ltd. Recipient(s) takes 
responsibility for any actions taken as a result of advice and recommendations 
given by Kesher Communications Ltd.



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[asterisk-users] Same extension registering over eth0 and eth1

[Zeeshan Zakaria]

Hello list,

I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.

Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only from
eth0, and if this port fails, sends registration coming in from eth1?

Zeeshan A Zakaria

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Re: [asterisk-users] Recording

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 18, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording

 

Hi,

 

Does anyone have a professional recording of someone saying "Recording" so I
can let the operator know that the one-touch recording has started
successfully?

 

Thanks

Dan

 

Just use SOX to rip the word out of one of the vm-* prompts.

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Re: [asterisk-users] Recording

[Dan Journo]

I would but it doesnt say "recording" at the start of the sentence, so the tone 
sounds wrong.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 18 October 2010 20:42
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 18, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording

Hi,

Does anyone have a professional recording of someone saying "Recording" so I 
can let the operator know that the one-touch recording has started successfully?

Thanks
Dan

Just use SOX to rip the word out of one of the vm-* prompts.
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Re: [asterisk-users] CEL Documentation

[Paul Belanger]

On Mon, Oct 18, 2010 at 2:54 PM, Jeremy Betts  wrote:
> Anybody know where to find some good information on the new CEL in
> asterisk 1.8? I'm very anxious to check out the new logging features but
> can't find anything but the cel.conf.sample file in the source package.
> I'd like to get this setup with ODBC.
>
It is very simple to get up and running. Especially if you already
have ODBC working.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Recording

[Dan Journo]

Thanks.
Thats perfect!
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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Paul Belanger]

On Mon, Oct 18, 2010 at 3:40 PM, Zeeshan Zakaria  wrote:
> Is there a way/software which can act as a middle man between asterisk and
> the ethernet ports, and by default sends registrations to asterisk only from
> eth0, and if this port fails, sends registration coming in from eth1?
>
DNS SRV or a SIP proxy.

-- 
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Zeeshan Zakaria]

Will OpenSIPs do the job?

Zeeshan A Zakaria

--
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On 2010-10-18 4:43 PM, "Paul Belanger"  wrote:

On Mon, Oct 18, 2010 at 3:40 PM, Zeeshan Zakaria  wrote:
> Is there a way/softwa...
DNS SRV or a SIP proxy.

--
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blog.polybeacon.com

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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Paul Belanger]

On Mon, Oct 18, 2010 at 4:47 PM, Zeeshan Zakaria  wrote:
> Will OpenSIPs do the job?
>
Any proxy would work, however I would re think your network design.

Re-registering the same phone, with the same extension, on the same
PBX is asking for trouble.  If you want to do redundancy, I would set
your network so only one ethernet route is active at one time, then it
is a matter or routing.  If you want both ethernet ports active, then
you are doing load balancing.  Something Asterisk by itself is not
strong at.  Hence the SIP proxy or DNS SRV records.

-- 
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blog.polybeacon.com

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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Zeeshan Zakaria]

I didn't design the network, it was already here at clien't site. It is
designed for redundancy. I am trying to come up with a solution to make
asterisk work in it. I am looking into opensips how it can help me.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-10-18 5:00 PM, "Paul Belanger"  wrote:

On Mon, Oct 18, 2010 at 4:47 PM, Zeeshan Zakaria  wrote:
> Will OpenSIPs do the ...
Any proxy would work, however I would re think your network design.

Re-registering the same phone, with the same extension, on the same
PBX is asking for trouble.  If you want to do redundancy, I would set
your network so only one ethernet route is active at one time, then it
is a matter or routing.  If you want both ethernet ports active, then
you are doing load balancing.  Something Asterisk by itself is not
strong at.  Hence the SIP proxy or DNS SRV records.

--

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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger ...
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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Daniel Tryba]

On Mon, Oct 18, 2010 at 04:38:28PM -0400, Paul Belanger wrote:
> > Is there a way/software which can act as a middle man between asterisk and
> > the ethernet ports, and by default sends registrations to asterisk only from
> > eth0, and if this port fails, sends registration coming in from eth1?
> >
> DNS SRV or a SIP proxy.

DNS SRV will not work since there is only 1 IP in this hot failover
network. A sip proxy introduces a single point of failure in the redundant 
network,
why not keep asterisk the single point of failure?

IMHO this kind of hot failover setup don't work with UDP traffic, it's
stateless. There where a duplicate TCP packet will be ignored, UDP will
be processed twice. While I didn't have this problem with SIP/RTP in a
hot failover network as described, we changed a hot setup to cold standby 
because of troubles with UDP traffic (can't remember the specifics).

-- 

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Re: [asterisk-users] warning diego viola the trouble maker for the world

[Diego Viola]

Hello everyone.

This person called Meftah Tayeb is saying all these things about me
because they kicked him out of his job at OVETEL, the truth is that he
wanted to make business with me pretending that he was the OVETEL
owner. And then I contacted the real owner and vice-president of
OVETEL and they told me that:

==

fromYaro Donchenko 
to  Diego Viola 
dateWed, Oct 13, 2010 at 7:18 PM
subject Re: [Rates] Please tell Meftah Tayeb to stop harass and damage
my company
mailed-by   ovetel.com

Diego,
I'm really sorry for this incident, the person is no longer related to OVETEL.

Thank You very much for notifying us, and once again, I'm very sorry for this.

Please let me know if I can do anything there.

More over he was freeswitch consultant, and not even close to be an owner.

Once again I'm very sorry for this and thank you very much for notifying us.

-- 
Best Regards
Yaro Donchenko
VP of Business Development

Direct: +1-845-475-9347
Mobile: +1-917-267-9276
MSN: y...@ovetel.com
Yahoo: yaroovetel

==

fromy...@ovetel.com 
to  diego.vi...@gmail.com
dateWed, Oct 13, 2010 at 8:05 PM
subject Chat with y...@ovetel.com
mailed-by   ovetel.com

7:49 PM Yaro: I'm really sorry to bother you
  can you please forward us entiere chat log you had
  and thank you very much one more time for notifying us about this

Yaro: the issue is we had open for him email long long time a go
  and then closed
  i tought
Yaro: but appertnly was not closed
7:51 PM he was at one point freeswitch consulant
  then he was doing some independant development for us in Algeria but
it did not take it any where
  so belive me

==

fromta...@ovetel.com 
to  diego.vi...@gmail.com
dateFri, Sep 10, 2010 at 12:50 PM
subject Chat with ta...@ovetel.com
mailed-by   ovetel.com

12:48 PM tayeb.meftah: http://www.ovetel.com
  my company
12:49 PM me: nice
  are you the owner of this company?
tayeb.meftah: me and other one

==

He used the name of a serious company, pretending he was the owner,
and asked me to send him money. OVETEL authorities apologized for the
incident, as you can see in the email above.

They have blocked his email, he used the one he got in the company. As
the Vice President of OVETEL said, he would no longer be part of the
company for trying to initiate a fraudulent deal.

I attached the conversations I had with the owners of OVETEL so you
can see who tells the truth and realize what kind of person Meftah
Tayeb is. Now he is trying to defame my company, Bridgecom LLC, which
is a legal company established in my country with all the legalities
involved.

Now that he doesn't even have a place at OVETEL, he spreads false
information about my company under false names. I want to state that I
have never asked him to do anything for free, actually, he asked me to
develop an application to manage DIDs for him for no cost.

I actually feel sorry for him, as he seems to think that everyone is
obliged to do him favors just because he's blind, at least he says he
is.

Nothing more to say, sincerely yours,

Diego Viola
Representative of Bridgecom LLC

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Re: [asterisk-users] CEL Documentation

[Jeremy Betts]

Thanks guys,

I think I've got enough to go on, I've setup the ODBC stuff before so I
think I should be able to figure this out.
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Re: [asterisk-users] a2billing

[Baha @ SH]

I am sorry , but where is System Settings??? And what is the parameter name?

And also, id like to mention that the voucher is working, only when balance
is below minimum balance it does not go to voucher ivr.

 

Thanks, awaiting,,,

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, October 18, 2010 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] a2billing

 

Turn on the voucher feature in System Settings and it will tell the user
right after the PIN authentication or CLID authentication that their balance
is below threshold and they should pay.

 

-Bruce

On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:

Not sure if a2billing can be shared here, but ill give a shot

If the credit < min_credit the IVR play: sorry you have 0 credit and hangup,
I want it to FW me to the IVR to add voucher, please let me know: here is
log:

[18/10/2010 07:01:12]:[file:a2billing.php -
line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
[18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE
: standard]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
[18/10/2010 07:01:12]:[file:a2billing.php -
line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
activated, inuse, simultaccess, typepaid, creditlimit, language,
removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
tariff=cc_tariffgroup.id WHERE username='9971524976']
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
language=en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
prepaid-zero-balance (cardnumber:9971524976)]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
callingcard_ivr_authenticate]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
(cia_res:-2)]]
[18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]





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[asterisk-users] IAX2 works one direction, but not the other...

[Cassius Smith]

I'm having trouble getting an IAX2 connection between a couple of
servers. I
can make calls from server B to server A, but when I call from Server A
to server
B, I get "No authority found".

If I remove serverA's password on ServerB's iax.conf, calls will go
through as "UNAUTHENTICATED".

On ServerA I am running Asterisk 1.6.2.9
On ServerB I'm running 1.6.2.13

Any hints for me?
The registrations in both directions seem to work fine when I do an iax2
reload from the CLI.

config file snips shown below.
Thanks
Cassius Smith
=

On server B, I have the following:
[general]
register => serverB:longsecretpasswo...@servera_ip

[serverA]
type=friend
host=dynamic
auth=md5
secret=longsecretpassword1
context=no911

[serverB]
type=friend
host=dynamic
auth=md5
secret=longsecretpassword2 ; if I remove this, calls go through as
UNAUTHENTICATED
context=no911

On server A, I have the following:
[general]
register => serverA:longsecretpasswo...@serverb_ip

[serverB]
type=friend
host=dynamic
auth=md5
secret=longsecretpassword2
context=no911

[cary]
type=friend
host=dynamic
auth=md5
secret=longsecretpassword1
context=no911


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[C F]

Ah Sandman http://sandman.com use a relay that goes onto an fxs port,
call that fxs port and you have a connection. Since that only work
momentary you will need a flip flop relay, the advantage is that by
calling it again you can turn it off.
Ring relay:
http://sandman.com/wizard.html#UniversalRingRelay
flip flop relay:
http://altronix.com/index.php?pid=2&model_num=RBR1224


On Mon, Oct 18, 2010 at 7:09 AM, Gilles  wrote:
> Hello
>
> I'm sure someone has already tried this: I use a couple of electric
> heaters to heat my office.
>
> I'd like to somehow connect them to Asterisk so that I could switch
> them on remotely by either calling the IVR or sending an e-mail to the
> Asterisk host, so that the room is warm when I get to the office :-)
>
> Any information appreciated.
>
> Thank you.
>
>
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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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[asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the fifth release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc5 is currently
scheduled to become the full release of Asterisk 1.8.0.

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform
compatibility IPv6 changes. In addition, the availability of the English sound
prompts with Australian accents has been added.

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[C F]

The other way is to use an RC3 from vikingelectronics.com
http://www.vikingelectronics.com/products/view_product.php?pid=217


On Mon, Oct 18, 2010 at 7:53 PM, C F  wrote:
> Ah Sandman http://sandman.com use a relay that goes onto an fxs port,
> call that fxs port and you have a connection. Since that only work
> momentary you will need a flip flop relay, the advantage is that by
> calling it again you can turn it off.
> Ring relay:
> http://sandman.com/wizard.html#UniversalRingRelay
> flip flop relay:
> http://altronix.com/index.php?pid=2&model_num=RBR1224
>
>
> On Mon, Oct 18, 2010 at 7:09 AM, Gilles  wrote:
>> Hello
>>
>> I'm sure someone has already tried this: I use a couple of electric
>> heaters to heat my office.
>>
>> I'd like to somehow connect them to Asterisk so that I could switch
>> them on remotely by either calling the IVR or sending an e-mail to the
>> Asterisk host, so that the room is warm when I get to the office :-)
>>
>> Any information appreciated.
>>
>> Thank you.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> 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] Asterisk 1.8.0 Release Candidate 5 Now Available

[Asterisk Development Team]

On 10-10-18 07:54 PM, Asterisk Development Team wrote:
> For a full list of changes in the current release candidate, please see the
> ChangeLog:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4

Apologies, this link should be:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5

-- The Asterisk Development Team

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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Bryant Zimmerman]

I would look at x10 triggered switches. There are some command line tools 
you could call from an IVR. 
I did a lot of x10 development on windows back in the day. I have seen some 
things for linux as well.

http://www.heyu.org/

Bryant


 From: "C F" 
Sent: Monday, October 18, 2010 7:55 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Asterisk to switch on electric heaters 
remotely?

Ah Sandman http://sandman.com use a relay that goes onto an fxs port,
call that fxs port and you have a connection. Since that only work
momentary you will need a flip flop relay, the advantage is that by
calling it again you can turn it off.
Ring relay:
http://sandman.com/wizard.html#UniversalRingRelay
flip flop relay:
http://altronix.com/index.php?pid=2&model_num=RBR1224

On Mon, Oct 18, 2010 at 7:09 AM, Gilles  wrote:
> Hello
>
> I'm sure someone has already tried this: I use a couple of electric
> heaters to heat my office.
>
> I'd like to somehow connect them to Asterisk so that I could switch
> them on remotely by either calling the IVR or sending an e-mail to the
> Asterisk host, so that the room is warm when I get to the office :-)
>
> Any information appreciated.
>
> Thank you.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> 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] a2billing

[bruce bruce]

I don't think voucher can be triggered to announce at certain threshold ONLY
but it will be run everytime at the begining after PIN is asked for. By
default it's set to: "Press 8 to fill up with a voucher".

System Settings is the last in the menu.

-Bruce

On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH  wrote:

> I am sorry , but where is System Settings??? And what is the parameter
> name?
>
> And also, id like to mention that the voucher is working, only when balance
> is below minimum balance it does not go to voucher ivr.
>
>
>
> Thanks, awaiting,,,
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Monday, October 18, 2010 12:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] a2billing
>
>
>
> Turn on the voucher feature in System Settings and it will tell the user
> right after the PIN authentication or CLID authentication that their balance
> is below threshold and they should pay.
>
>
>
> -Bruce
>
> On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:
>
> Not sure if a2billing can be shared here, but ill give a shot
>
> If the credit < min_credit the IVR play: sorry you have 0 credit and
> hangup,
> I want it to FW me to the IVR to add voucher, please let me know: here is
> log:
>
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:76]:[CallerID:]:[CN:]:[MODE
> : standard]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
> SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
> activated, inuse, simultaccess, typepaid, creditlimit, language,
> removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
> expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
> UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
> cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
> cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
> tariff=cc_tariffgroup.id WHERE username='9971524976']
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
> active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
> language=en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
> prepaid-zero-balance (cardnumber:9971524976)]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
> callingcard_ivr_authenticate]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
> (cia_res:-2)]]
> [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]
>
>
>
>
>
> --
> _
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>
>
>
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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Cassius Smith]

BTW I apologize for the double send. 




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Re: [asterisk-users] Same extension registering over eth0 and eth1

[Philipp von Klitzing]

Hi!

> Is there a way/software which can act as a middle man between asterisk
> and the ethernet ports, and by default sends registrations to asterisk
> only from eth0, and if this port fails, sends registration coming in
> from eth1? 

Spanning Tree (STP, RSTP, MSTP)


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Re: [asterisk-users] a2billing

[Baha @ SH]

Exactly,

I don't want that, it's annoying! I just want it to run if the customer
balance reach for example < 1 dollar!

Anyway?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, October 18, 2010 8:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] a2billing

 

I don't think voucher can be triggered to announce at certain threshold ONLY
but it will be run everytime at the begining after PIN is asked for. By
default it's set to: "Press 8 to fill up with a voucher".

 

System Settings is the last in the menu.

 

-Bruce

On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH  wrote:

I am sorry , but where is System Settings??? And what is the parameter name?

And also, id like to mention that the voucher is working, only when balance
is below minimum balance it does not go to voucher ivr.

 

Thanks, awaiting,,,

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, October 18, 2010 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] a2billing

 

Turn on the voucher feature in System Settings and it will tell the user
right after the PIN authentication or CLID authentication that their balance
is below threshold and they should pay.

 

-Bruce

On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:

Not sure if a2billing can be shared here, but ill give a shot

If the credit < min_credit the IVR play: sorry you have 0 credit and hangup,
I want it to FW me to the IVR to add voucher, please let me know: here is
log:

[18/10/2010 07:01:12]:[file:a2billing.php -
line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
[18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE
: standard]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
[18/10/2010 07:01:12]:[file:a2billing.php -
line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
activated, inuse, simultaccess, typepaid, creditlimit, language,
removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
tariff=cc_tariffgroup.id WHERE username='9971524976']
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
language=en]]
[18/10/2010 07:01:12]:[file:Class.A2Billing.php -
line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
prepaid-zero-balance (cardnumber:9971524976)]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
callingcard_ivr_authenticate]]
[18/10/2010 07:01:14]:[file:a2billing.php -
line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
(cia_res:-2)]]
[18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]





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Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available

[Barry Miller]

On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
> On 10-10-18 07:54 PM, Asterisk Development Team wrote:
> > For a full list of changes in the current release candidate, please see the
> > ChangeLog:
> >
> > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
> 
> Apologies, this link should be:
> 
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5
> 
> -- The Asterisk Development Team

Is it worth mentioning somewhere (ChangeLog? This list?) that all the
asterisk-core-sounds tarballs were updated today?  It would remind someone
[me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a
'make sounds' before stopping asterisk for the install.  My test system is
on a slow link, and waiting for the tarball downloads in the middle of
installing is frustrating.

-- 
Barry

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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Jigar Joshi]

I will have a closer look at this book

My Question is :

Is it possible with asterisk to resolve all the code to one extension and
with the extension no.

For example person one calls it and enters code 1
person two calls and enters code 2

both the call should received by a single extension say 1001.
and there I should be able to differenciate both the calls using code
entered.

in the example: both the call will be given to extension 1001 and at 1001
there will be an app running that will make this into two calls.i mean each
packet contains the code entered.

I hope this answer would be helpful .

Thanks.



On Mon, Oct 18, 2010 at 9:39 PM, Steve Edwards wrote:

> On Mon, 18 Oct 2010, Jigar Joshi wrote:
>
> > @Gilles here are my requirement.can you please help me .
>
> Are you putting this "out to bid" or are you just too lazy to read ATFOT
> (http://downloads.oreilly.com/books/9780596510480.pdf)?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> _
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Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Steve Edwards]

Un-top-posting...


  On Mon, 18 Oct 2010, Jigar Joshi wrote:

  > @Gilles here are my requirement.can you please help me .


On Mon, Oct 18, 2010 at 9:39 PM, Steve Edwards 
 wrote:


Are you putting this "out to bid" or are you just too lazy to read ATFOT 
(http://downloads.oreilly.com/books/9780596510480.pdf)?


On Tue, 19 Oct 2010, Jigar Joshi wrote:


I will have a closer look at this book

My Question is :

Is it possible with asterisk to resolve all the code to one extension 
and with the extension no.


For example person one calls it and enters code 1 person two calls and 
enters code 2  


both the call should received by a single extension say 1001. and there 
I should be able to differenciate both the calls using code entered.


in the example: both the call will be given to extension 1001 and at 
1001 there will be an app running that will make this into two calls.i 
mean each packet contains the code entered.


Reading the book will help you understand the terminology so you will 
either learn how to write a dialplan to suit your needs or ask questions 
that can be answered.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
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