date:20130531

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

2013-05-31 Thread Bob Kyeyune

hello;
hopefully u can help me
i have asterisk vanilla installation 11 and i have also managed to install
webrtc2sip
how do i make asterisk to communicate with webrtc2sip c'se right now both
run independently


Regards.
Kyeyune Bob
Network & IT Engineer
+256 774 702 258
bob.kyey...@onesolutions.ug

Integrated IT services from
 Plot 57B Luthuli Avenue Bugolobi, Kampala






On Fri, May 31, 2013 at 11:36 PM, Adnan <112linuxstockh...@gmail.com> wrote:

> Voxeo/Phono webrtc.
>
> /Adnan
>
>
> On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri wrote:
>
>>
>> Hi All,
>> I wonder if any of you has some suggestions on which WebRTC
>> client/softphone to use for a click-to-dial, webpage hosted solution. Any
>> suggestions?
>> Thanks
>> l.
>> --
>> Loway - home of QueueMetrics - http://queuemetrics.com
>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] How to know the conflict in the dependencies?

2013-05-31 Thread Jean-Denis Girard

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Le 31/05/2013 15:10, bilal ghayyad a écrit :
> Hello;
> 
> When I type make menuselect and finding the channels that has the sign XXX 
> before it (this at the driver), how can I know the dependencies that are 
> causing this conflict?

Dependencies are detailed at the bottom of menuselect screen, for example
  XXX chan_motif

   Motif Jingle Channel Driver
   Depends on: iksemel(E), res_xmpp(M)
   Can use: openssl(E)

   Support Level: core



Thanks,
- -- 
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
-BEGIN PGP SIGNATURE-

iEYEARECAAYFAlGpT0IACgkQuu7Rv+oOo/hkcwCgrMjVmIu1ftcMi9T1HSa6wTpZ
upMAnixIgG4wYusuf02/zp9EByUgQcJY
=Quzq
-END PGP SIGNATURE-

--
_
-- 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

[asterisk-users] How to know the conflict in the dependencies?

2013-05-31 Thread bilal ghayyad

Hello;

When I type make menuselect and finding the channels that has the sign XXX 
before it (this at the driver), how can I know the dependencies that are 
causing this conflict?

Regards
Bilal

--
_
-- 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

Re: [asterisk-users] Help me understand these log messages

2013-05-31 Thread Chris Gentle

OK, I understand now.  I didn't realize allowguest was on by default.
I guess I should read more closely.  Thanks!

On Fri, May 31, 2013 at 5:15 PM, Yves A.  wrote:
> ... an anonyous (not registerted) sip user from 188.161.238.232 was trying
> to initiate a call to
> 9725955 and so on...
> you could enable sip tracing to get more information.
>
> maybe you should change the 'allowguest' option in sip.conf..?
>
> regards,
> yves
>
> Am 31.05.2013 23:57, schrieb Chris Gentle:
>>
>> OK, I need a bit of help here.  I'm configuring a new Asterisk 11
>> system and I accidentally let my firewall rules drop for a day or so.
>> When I logged in today, I found messages like the ones below on my
>> asterisk console.  Obviously somebody was trying to take advantage of
>> my carelessness.  So can someone explain what would cause these types
>> of messages to show up on my console?
>>
>> I understand that my iptables would have stopped this but I'm just
>> trying to understand more about the problem.  What other settings
>> might have stopped this?  Fail2ban was running but there were no
>> "failed registration" type messages that would have triggered it.
>>
>> [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
>> (188.161.238.232:28203) to extension '972595595767' rejected because
>> extension not found in context 'default'.
>> [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
>> RTP CoS mark 5
>> [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
>> (188.161.238.232:28203) to extension '00972595595767' rejected because
>> extension not found in context 'default'.
>> [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
>> RTP CoS mark 5
>> [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
>> (188.161.238.232:28203) to extension '000972595595767' rejected
>> because extension not found in context 'default'.
>> [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
>> RTP CoS mark 5
>> [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
>> (188.161.238.232:28203) to extension '011972595595767' rejected
>> because extension not found in context 'default'.
>> 
>>
>>
>> --
>> Chris
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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



-- 
Chris

--
_
-- 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

Re: [asterisk-users] Help me understand these log messages

2013-05-31 Thread Alec Davis

Top of sip.conf

;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
;   understand the risks of installing Asterisk with the sample
;   configuration. If your Asterisk is installed on a public
;   IP address connected to the Internet, you will want to learn
;   about the various security settings BEFORE you start
;   Asterisk.
;
;   Especially note the following settings:
;   - allowguest (default enabled)
;   - permit/deny/acl - IP address filters
;   - contactpermit/contactdeny/contactacl - IP address filters
for registrations
;   - context - Which set of services you offer various users
;


In other words: allowguest = yes, is the default.
But in trunk the context for guest is [public], yours started in the
[default] context

Alec 

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> Chris Gentle
> Sent: Saturday, 1 June 2013 9:57 a.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Help me understand these log messages
> 
> OK, I need a bit of help here.  I'm configuring a new 
> Asterisk 11 system and I accidentally let my firewall rules 
> drop for a day or so.
> When I logged in today, I found messages like the ones below 
> on my asterisk console.  Obviously somebody was trying to 
> take advantage of my carelessness.  So can someone explain 
> what would cause these types of messages to show up on my console?
> 
> I understand that my iptables would have stopped this but I'm 
> just trying to understand more about the problem.  What other 
> settings might have stopped this?  Fail2ban was running but 
> there were no "failed registration" type messages that would 
> have triggered it.
> 
> [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
> (188.161.238.232:28203) to extension '972595595767' rejected 
> because extension not found in context 'default'.
> [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
> RTP CoS mark 5
> [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
> (188.161.238.232:28203) to extension '00972595595767' 
> rejected because extension not found in context 'default'.
> [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
> RTP CoS mark 5
> [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
> (188.161.238.232:28203) to extension '000972595595767' 
> rejected because extension not found in context 'default'.
> [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
> RTP CoS mark 5
> [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
> (188.161.238.232:28203) to extension '011972595595767' 
> rejected because extension not found in context 'default'.
> 
> 
> 
> --
> Chris
> 
> --
> _
> -- 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


--
_
-- 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

Re: [asterisk-users] Help me understand these log messages

2013-05-31 Thread Yves A.

... an anonyous (not registerted) sip user from 188.161.238.232 was 
trying to initiate a call to

9725955 and so on...
you could enable sip tracing to get more information.

maybe you should change the 'allowguest' option in sip.conf..?

regards,
yves

Am 31.05.2013 23:57, schrieb Chris Gentle:

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
"failed registration" type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.



--
Chris

--
_
-- 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




--
_
-- 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

[asterisk-users] Help me understand these log messages

2013-05-31 Thread Chris Gentle

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
"failed registration" type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.



--
Chris

--
_
-- 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

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

2013-05-31 Thread Adnan

Voxeo/Phono webrtc.

/Adnan


On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri  wrote:

>
> Hi All,
> I wonder if any of you has some suggestions on which WebRTC
> client/softphone to use for a click-to-dial, webpage hosted solution. Any
> suggestions?
> Thanks
> l.
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] TE410P PCI card in 1U rackount server

2013-05-31 Thread Andre Goree

On 05/31/2013 03:42 PM, jg wrote:
> I have some 1-U servers running using Xeon processors with the Asus
> RS100-E7/PI2 barebone. There's only room for a single card and in my
> case I am using PCI-E Sangoma B500 cards.  Geometry is the basic
> problem. But this should not be a problem for your T1/E1 card. I had
> problems using a Sangoma B700 hybrid card. You do need an adapter (if it
> doesn't come with the board or barebone) as the cards are at least of
> 2-U size.
> 
> jg
> 

Thanks!  Yeah I'll certainly need the adapter since one did not come
with the board, I guess I was looking for advice on which one's people
are using with their cards, but I suppose and ol' pci-e -> 32-bit PCI
should work in the case of the TE410P, lol.


-- 
Andre Goree
Linux Systems Administrator, Atlantic.Net
Toll Free: 1.800.540.4686
International: +1.321.206.3731
Mobile:  407.272.6886

--
_
-- 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

Re: [asterisk-users] TE410P PCI card in 1U rackount server

2013-05-31 Thread jg

I have some 1-U servers running using Xeon processors with the Asus 
RS100-E7/PI2 barebone. There's only room for a single card and in my 
case I am using PCI-E Sangoma B500 cards.  Geometry is the basic 
problem. But this should not be a problem for your T1/E1 card. I had 
problems using a Sangoma B700 hybrid card. You do need an adapter (if it 
doesn't come with the board or barebone) as the cards are at least of 
2-U size.


jg

--
_
-- 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

[asterisk-users] TE410P PCI card in 1U rackount server

2013-05-31 Thread Andre Goree

I was wondering if there is anyone following the list that has this or
similar PCI cards running in a 1U server.  If so, could you possibly
recommend a PCI card bus adapter?  I'm not sure exactly which kind I'll
need.

I plan on researching the different types of adapters I can purchase for
my specific mobo (Supermicro X9SCA-F), but I'd like to not waste time by
ordering an incorrect adapter and figured it might be prudent to ask you
good folks :)

Thanks in advance for any guidance you can provide.

-- 
Andre Goree
Linux Systems Administrator, Atlantic.Net
Toll Free: 1.800.540.4686
International: +1.321.206.3731
Mobile:  407.272.6886

--
_
-- 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

Re: [asterisk-users] how to launch a URl when dialing a number

2013-05-31 Thread Warren Selby

On Fri, May 31, 2013 at 11:29 AM, Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:

> hello ,
>
> thanks alex for your help and support the scenario is correct.
>
> i will try to follow your suggestion and i will update you asap
>
> thank you again for your explication i really appreciate it
>
>

Have you tried maybe setting up the entire call in an AGI that will execute
the desired script as you make the dial command?  Or, you could look at
running the M or U options in your Dial() command to execute a macro or
gosub routine when the call is connected?

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
--
_
-- 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

Re: [asterisk-users] how to launch a URl when dialing a number

2013-05-31 Thread A J Stiles

On Friday 31 May 2013, Alex Villacís Lasso wrote:
>  From this discussion, I am guessing the following scenario. Please correct
> me if I am wrong. - There are (at least) three roles in your scenario: the
> Asterisk server, the PHP webserver (which may or may not be the same
> machine as the Asterisk server), and the client PC. - Apparently your
> client PC runs a softphone (but the exact nature of the telephony client
> is not important). - A call is connected from the phone to your Asterisk,
> is directed to your context, and dials some trunk (Zap/g1 in your
> snippet). - You then want, somehow, to make the Asterisk server reach out
> to your client PC (which runs a GUI and has a web browser) and force it to
> open an arbitrary web page on the PHP webserver, presumably a callcenter
> data collecting form.
> 
> The problematic issue is the last part. Especially the implication of
> remotely opening a web page on some random PC.
> 
> If the above scenario is in fact what you were planning to do, maybe you
> need to rethink your design. In the default case, there is no way to make
> a remote PC open an arbitrary URL on its GUI. Think about the security
> implications. You should instead have the web interface already open, and
> program a Click2Call capability that contacts the Asterisk server and uses
> AMI to execute an Originate action with your context as your target. Then
> the web page would load your target URL in order to handle the call. Or,
> if the calls come from an external source, you should program some kind of
> monitor that alerts the web interface that the call was handled by the
> context.

Actually, that *is* doable, because I have done it!

I wrote a little daemon, which runs on each agent's workstation, listening for 
UDP messages and delivering system notifications.  The daemon is started 
through a symlink in /home/*/.kde/Autostart, so it runs under the logged-in 
user's ID; and one of the supported message types instructs the daemon to open 
a URL in iceweasel.  Since iceweasel probably is already running, it just 
opens a new tab in the existing browser process.

This isn't quite as insecure as it sounds, because I made sure to bake in the 
following security measures:

*  You can't actually execute arbitrary commands just by sending UDP messages.  
The daemon is doing some sanity-checking -- the worst you can do is display an 
arbitrary web page.

*  All this is happening on the inside of a firewall, which blocks access from 
agents' workstations to most outside IP addresses.  Any web page you can 
display is safe-for-work.

*  The firewall also doesn't let anything in on the UDP port which is used for 
notification messaging.

When an agent receives an incoming call on their DDI number, an AGI script is 
started.  It looks up the calling number in the database; and if it is found, 
a notification message is sent with the details of the caller, followed by 
another one to fetch up the user's details in our web app.

There is no reason why an AGI script should not fire off a request to open a 
browser page as an external number is dialled.  But instead of dialling the 
number manually, we use a callfile-generating CGI script  (linked from within 
the main telesales application)  to call external numbers when the agent 
follows a link.

It probably also helps greatly that all the software we use to run our call 
centre  (besides whatever is usually found in a standard Linux distribution)  
was written in-house .

-- 
AJS

Answers come *after* questions.

--
_
-- 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

Re: [asterisk-users] how to launch a URl when dialing a number

2013-05-31 Thread Salaheddine Elharit

hello ,

thanks alex for your help and support the scenario is correct.

i will try to follow your suggestion and i will update you asap

thank you again for your explication i really appreciate it


2013/5/31 Alex Villacís Lasso 

>  El 31/05/13 09:21, Salaheddine Elharit escribió:
>
>  thanks justin i try to do this but the issue still the same.this link is
> stored in my server 192.168.5.109 .but what i want to receive this link
> when i call this number in my pc
>
>  ip adresse of my pc 192.168.5.131
> ip adresse of server when the page php is stored
>
>  thanks and regards
>
>
>
>  2013/5/30 Justin Killen 
>
>>  If you just want the url to be opened (perhaps to update a counter via
>> a web service or cgi script), you can do this:
>>
>>
>>
>> system(“wget http://” )
>>
>> or
>>
>> system(“fetch http://...” )
>>
>>
>>
>>
>>
>>
>>
>> -Justin
>>   --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
>> Elharit
>> *Sent:* Thursday, May 30, 2013 8:07 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] how to launch a URl when dialing a number
>>
>>
>>
>> Hello
>>
>>
>>
>> i want to luanch an URL in my PC when i call a number  like below
>>
>>
>>
>> exten => 066104,1,Set(CALLERID(number)=52xxx)
>>
>> exten
>> => 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>>
>> exten
>> => 
>> 066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
>>
>> exten => 066104,n,http://192.168.5.109/interface2/interface2.php (
>> here i want to launch this url in my pc )
>>
>> exten => 066104,n,Hangup()
>>
>>
> From this discussion, I am guessing the following scenario. Please correct
> me if I am wrong.
> - There are (at least) three roles in your scenario: the Asterisk server,
> the PHP webserver (which may or may not be the same machine as the Asterisk
> server), and the client PC.
> - Apparently your client PC runs a softphone (but the exact nature of the
> telephony client is not important).
> - A call is connected from the phone to your Asterisk, is directed to your
> context, and dials some trunk (Zap/g1 in your snippet).
> - You then want, somehow, to make the Asterisk server reach out to your
> client PC (which runs a GUI and has a web browser) and force it to open an
> arbitrary web page on the PHP webserver, presumably a callcenter data
> collecting form.
>
> The problematic issue is the last part. Especially the implication of
> remotely opening a web page on some random PC.
>
> If the above scenario is in fact what you were planning to do, maybe you
> need to rethink your design. In the default case, there is no way to make a
> remote PC open an arbitrary URL on its GUI. Think about the security
> implications. You should instead have the web interface already open, and
> program a Click2Call capability that contacts the Asterisk server and uses
> AMI to execute an Originate action with your context as your target. Then
> the web page would load your target URL in order to handle the call. Or, if
> the calls come from an external source, you should program some kind of
> monitor that alerts the web interface that the call was handled by the
> context.
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] how to launch a URl when dialing a number

2013-05-31 Thread Alex Villacís Lasso


El 31/05/13 09:21, Salaheddine Elharit escribió:

thanks justin i try to do this but the issue still the same.this link is stored 
in my server 192.168.5.109 .but what i want to receive this link when i call 
this number in my pc

ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored

thanks and regards



2013/5/30 Justin Killen mailto:jkil...@allamericanasphalt.com>>

If you just want the url to be opened (perhaps to update a counter via a 
web service or cgi script), you can do this:

system("wget http://";)

or

system("fetch http://...";)

-Justin




*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com 
] *On Behalf Of *Salaheddine Elharit
*Sent:* Thursday, May 30, 2013 8:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] how to launch a URl when dialing a number

Hello

i want to luanch an URL in my PC when i call a number  like below

exten => 066104,1,Set(CALLERID(number)=52xxx)

exten => 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))

exten => 
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))

exten => 066104,n,http://192.168.5.109/interface2/interface2.php ( here 
i want to launch this url in my pc )

exten => 066104,n,Hangup()




From this discussion, I am guessing the following scenario. Please correct me 
if I am wrong.
- There are (at least) three roles in your scenario: the Asterisk server, the 
PHP webserver (which may or may not be the same machine as the Asterisk 
server), and the client PC.
- Apparently your client PC runs a softphone (but the exact nature of the 
telephony client is not important).
- A call is connected from the phone to your Asterisk, is directed to your 
context, and dials some trunk (Zap/g1 in your snippet).
- You then want, somehow, to make the Asterisk server reach out to your client 
PC (which runs a GUI and has a web browser) and force it to open an arbitrary 
web page on the PHP webserver, presumably a callcenter data collecting form.

The problematic issue is the last part. Especially the implication of remotely 
opening a web page on some random PC.

If the above scenario is in fact what you were planning to do, maybe you need to rethink your design. In the default case, there is no way to make a remote PC open an arbitrary URL on its GUI. Think about the security implications. You should instead have 
the web interface already open, and program a Click2Call capability that contacts the Asterisk server and uses AMI to execute an Originate action with your context as your target. Then the web page would load your target URL in order to handle the call. 
Or, if the calls come from an external source, you should program some kind of monitor that alerts the web interface that the call was handled by the context.
--
_
-- 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

Re: [asterisk-users] how to launch a URl when dialing a number

2013-05-31 Thread Salaheddine Elharit

thanks justin i try to do this but the issue still the same.this link is
stored in my server 192.168.5.109 .but what i want to receive this link
when i call this number in my pc

ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored

thanks and regards



2013/5/30 Justin Killen 

> **
>
> If you just want the url to be opened (perhaps to update a counter via a
> web service or cgi script), you can do this:
>
> ** **
>
> system(“wget http://”)
>
> or
>
> system(“fetch http://...”)
>
> ** **
>
> ** **
>
> ** **
>
> -Justin 
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine
> Elharit
> *Sent:* Thursday, May 30, 2013 8:07 AM
> *To:* **Asterisk Users Mailing List - Non-Commercial Discussion**
> *Subject:* [asterisk-users] how to launch a URl when dialing a number
>
> ** **
>
> Hello 
>
> ** **
>
> i want to luanch an URL in my PC when i call a number  like below
>
> ** **
>
> exten => 066104,1,Set(CALLERID(number)=52xxx)
>
> exten => 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> 
>
> exten
> => 
> 066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
> 
>
> exten => 066104,n,http://192.168.5.109/interface2/interface2.php (
> here i want to launch this url in my pc )
>
> exten => 066104,n,Hangup() 
>
> ** **
>
> ** **
>
> thanks and regards
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] G.729 codec in pass-thru mode

2013-05-31 Thread Matthew J. Roth

Kamlesh Kumar wrote:
> 
> Yes that's correct, when I use u-law call works fine.
> 
> In case of g729, I enabled sip debug with 'sip set debug on' and captured all
> the sip traces and got whatever I posted in last email. There was no other
> call on the system when I captured sip trace. Please suggest further
> proceedings. 


Kamlesh,

Please provide a SIP trace (sip set debug on) of a successful u-law call.  I'm
especially interested in the dialog between the Asterisk server and the ITSP in
this scenario.

Also include the relevant sections of sip.conf and the dialplan.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

--
_
-- 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

Re: [asterisk-users] G.729 codec in pass-thru mode

2013-05-31 Thread Kamlesh Kumar

Matthew,
 
Yes that's correct, when I use u-law call works fine. 
 
In case of g729, I enabled sip debug with 'sip set debug on' and captured all 
the sip traces and got whatever I posted in last email. There was no other call 
on the system when I captured sip trace. Please suggest further proceedings.
 
Regards,
Kamlesh
 
> Date: Wed, 29 May 2013 08:42:39 -0500
> From: mr...@imminc.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
> 
> Kamlesh Kumar wrote:
> >
> > Call even doesn't go to the ITSP. I tried without AGI script and the results
> > were same.
> 
> 
> Kamlesh,
> 
> Your first message stated that the call is successful if the codec is u-law, 
> so
> there must be communication between the Asterisk server and the ITSP.  The key
> to understanding why the G.729 call fails is in this SIP signaling.
> 
> How are you capturing the SIP trace?  Are you enabling SIP debugging for the
> specific SIP softphone?  If so, please use "sip set debug on" to enable it for
> all SIP packets.  Then wait until there are no other calls on the Asterisk
> server, try another G.729 call, and post the CLI output.
> 
> Regards,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> --
> _
> -- 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
  --
_
-- 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

[asterisk-users] WebRTC softphone for Asterisk - any suggestion?

2013-05-31 Thread Lenz Emilitri

Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted solution. Any
suggestions?
Thanks
l.
-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
--
_
-- 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

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

Re: [asterisk-users] How to know the conflict in the dependencies?

[asterisk-users] How to know the conflict in the dependencies?

Re: [asterisk-users] Help me understand these log messages

Re: [asterisk-users] Help me understand these log messages

Re: [asterisk-users] Help me understand these log messages

[asterisk-users] Help me understand these log messages

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

Re: [asterisk-users] TE410P PCI card in 1U rackount server

Re: [asterisk-users] TE410P PCI card in 1U rackount server

[asterisk-users] TE410P PCI card in 1U rackount server

Re: [asterisk-users] how to launch a URl when dialing a number

Re: [asterisk-users] how to launch a URl when dialing a number

Re: [asterisk-users] how to launch a URl when dialing a number

Re: [asterisk-users] how to launch a URl when dialing a number

Re: [asterisk-users] how to launch a URl when dialing a number

Re: [asterisk-users] G.729 codec in pass-thru mode

Re: [asterisk-users] G.729 codec in pass-thru mode

[asterisk-users] WebRTC softphone for Asterisk - any suggestion?

19 matches

Site Navigation

Mail list logo

Footer information