[asterisk-users] missed call notification

[tux john]

Hi. i am running asterisk 11 in debian and i would like have a missed call notification down to extension level.

so if i get a missed call to extension 6589 then send an email to the user's email address with a subject and a text message.

is there a guide on how to create something like that?

 

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Re: [asterisk-users] iaxmodem errors.

[tux john]

ID(name)}, Tel: ${CALLERID(num)});
          .
          . other conditions checked and extensions dialled
          .
          };
  
          fax => {
                  NoOp(Fax Detected ${STRFTIME(,,%F %T %z)});
                  Set(FAXOPT(gateway)=yes);
                  Queue(hylafax-iax,dRt,,,15);
  
                  Set(FAXOPT(gateway)=no);
                  &fax-receive(,,FaxMaster,lmoore);
                  Hangup();
          };
                 
          h => {
      if ( "X${FAXRXFILE}" != "X" )
      {
      &email_rxfax();
      }
      NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)});
      };
  };
  
  macro fax-receive( fax-number, header-info, sender, recipient ) {
  /*
      ${ARG1} is Receiving Station Fax Number
      ${ARG2} is Fax Header Information
      ${ARG3} is Fax Sender E-mail Address
      ${ARG4} is Fax Recipient E-mail Address
  */
      NoOp( FAX RECEIVE );
      Set(FAXOPT(localstationid)=${LOCAL(fax-number)});
      Set(FAXOPT(headerinfo)=${LOCAL(header-info)});
      Set(FROMADDR=${LOCAL(sender)});
      Set(TOADDR=${LOCAL(recipient)});
      NoOp( SETTING FAXOPT );
      NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});
      NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});
      NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)});
      Set(RXSTART=${EPOCH});
      Set(FAXRXPATH=/var/spool/asterisk/fax/received);
      Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});
      NoOp( RECEIVING FAX : ${FAXRXFILE} );
      ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);
      NoOp( Subroutine Return );
      return;
      };

Cheers,

Larry.



  On 13/11/2016 8:07 AM, Larry Moore wrote:


  Is your network/firewall configuration permitting the ports for UDPTL, runn the command:  udptl show config
  
  
UDPTL Global options

udptlstart:  4000
udptlend:    4999
udptlfecentries: 3
udptlfecspan:    3
use_even_ports:  No
udptlchecksums: Yes
  
  
  In your sip configuration for your 'mytrunk' peer have you set applicable options e.g.:
  
t38pt_udptl=yes,redundancy,maxdatagram=400
  
  In your extensions.conf you could and probably should set the following option prior to dialing the IAX channel, this is to enable the T.38 gateway feature of Asterisk 11:
  
Set(FAXOPT(gateway)=yes)
  
  I have it working in my installation however I have incoming voice calls too hence I use 'faxdetect' to direct the call to the 'fax' extension.
  
  Cheers,
  
  Larry.
  
  
On 12/11/2016 5:24 AM, tux john wrote:
  
  

  
hi. i am using asterisk 11.24.1 in my raspberry. i do have a sip trunk with a provider with g711a. I am trying to setup a fax server by following the guide in http://the-asterisk-book.com/1.6/faxserver.html.
  
  
 
  
  
i do live in Greece and the number is 00302112152130
  
  
the problem is that i am getting the following error and i am stuck:
  
  
 
  
  

    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
      -- Executing [00302112152130@fax-in:1] Dial("SIP/mytrunk-0001", "IAX2/iaxmodem") in new stack
      -- Called IAX2/iaxmodem
      -- Hungup 'IAX2/iaxmodem-3818'
    == Everyone is busy/congested at this time (1:0/0/1)
      -- Auto fallthrough, channel 'SIP/mytrunk-0001' status is 'CHANUNAVAIL'
  RasPBX*CLI>


   


   


  the extensions.conf has


  
[fax-in]
exten => 00302112152130,1,Dial(IAX2/iaxmodem)
  
  
 
  


   


  any ideas, please?

  



  
  
  
  


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[asterisk-users] iaxmodem errors.

[tux john]

hi. i am using asterisk 11.24.1 in my raspberry. i do have a sip trunk with a provider with g711a. I am trying to setup a fax server by following the guide in http://the-asterisk-book.com/1.6/faxserver.html.

 

i do live in Greece and the number is 00302112152130

the problem is that i am getting the following error and i am stuck:

 


  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [00302112152130@fax-in:1] Dial("SIP/mytrunk-0001", "IAX2/iaxmodem") in new stack
    -- Called IAX2/iaxmodem
    -- Hungup 'IAX2/iaxmodem-3818'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/mytrunk-0001' status is 'CHANUNAVAIL'
RasPBX*CLI>

 

 

the extensions.conf has


[fax-in]
exten => 00302112152130,1,Dial(IAX2/iaxmodem)

 


 

any ideas, please?


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Re: [asterisk-users] Feature access codes

[tux john]

I did edit the features.conf. what about extensions.conf?
On 16/10/2016, 14:45 Doug Lytle  wrote:

  

  On 10/16/2016 02:33 AM, tux john wrote:


  The problem is that I do not know how to configure the feature access codes including transfer.


Review features.conf

Doug

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Re: [asterisk-users] Feature access codes

[tux john]

Thanks for the reply. I do know the security practices and I am using VoIP. The problem is that I do not know how to configure the feature access codes including transfer.
On 15/10/2016, 21:42 Steve Edwards <asterisk.org@sedwards.com> wrote:

  
On Sat, 15 Oct 2016, tux john wrote:

> Hi. Kinda new to the area and I would like some help please. I have
> asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed
> to each user and 2 DIDs for faxing. Everything works fine but I do not
> have call transfer between extensions and feature access codes. I have
> read somewhere that enabling call transfer can be a security hole for
> sip attackers.

Are these incoming calls copper or VOIP?

If you only accept copper calls, make sure Asterisk is only listening to
127.0.0.1 and enforce this policy with another layer dropping any incoming
SIP packets at the firewall.

If you only intend to accept calls from your ISP, configure Asterisk to
only accept calls from your ISP, and enforce this policy at the firewall.

If you accept calls from everyone, re-think your definition of 'everyone.'
It probably does not include Iraq, North Korea, China, Russia, etc.
Configure Asterisk and your firewall accordingly.

Beyond this, follow 'best practices' (google for sip best practices --
John Todd did a list years back, Nerdvittles probably will also be a good
resource) like long, random user names and passwords, only allow needed
features to each class of users, etc.

--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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[asterisk-users] Feature access codes

[tux john]

Hi. Kinda new to the area and I would like some help please.  I have asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed to each user and 2 DIDs for faxing. Everything works fine but I do not have call transfer between extensions and feature access codes. I have read somewhere that enabling call transfer can be a security hole for sip attackers.

I would appreciate any help available, please.


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[asterisk-users] send a call to moh until user is available

[tux john]

in asterisk 11 i have an extension 4450 that gets all the calls from a particular number (eg 1234567890).

in extensions.conf i do have the following:


exten => 1234567890,n,Dial(SIP/4450,20,mg)

exten => 1234567890,n,Hangup()

 

the problem that i have is that when extension 4450 is busy then the call goes to hangup. i would like if the extension 4450 is busy the caller to hear moh and when available return to the extension 4450. I would like that to be for 1 minute, then busy (3) and then hangup.

could someone provide me an example, please?


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[asterisk-users] mysql phonebook

[tux john]

hi. i am running asterisk 11 and i am using astdb to store all my contacts and their numbers. so everytime they call me, i can see their name on the screen of the phone.
 i am making use of the following to retrieve the name from the astdb

exten => WhatEverIsMyDID,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => WhatEverIsMyDID,n,Answer()

 

in the same machine i have mysql. i would like to make use of mysql to store and retrieve phonebook as well create blacklist of numbers.

i thought of creating 2 databases

-phonebook, will contain name, number

-blacklist, will contain name, number

 

once i update all the database fiels how can i see the names whenever someone calls?

regarding the blacklist, i would like to send them to hear a sound (eg tt-monkeys) or simply hangup.

 

Some advice please?

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[asterisk-users] voicemail greeting

[tux john]

hi.i managed to record my voicemail greeting. the only problem is that after my greeting the caller hear '...please leave your message after the tone. when done press the pound key or hangup.' is there a way to get rid of that?

Ideally i would like to have my own recording and then the beep sound.


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[asterisk-users] Blacklist callers from file

[tux john]

Hi. I would like to blacklist a few callers and I have been using the *CLI> database put blacklist 1234 "annoying callers". Instead of putting the same command for every user is there any way to have a file? Ideally a file in /opt that I would update the blacklisted numbers (add,remove). Is there anything like that, please?

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[asterisk-users] Google contacts sync

[tux john]

Hi. I am running asterisk 11 without any GUI and I am running a phonebook by making use of asterisk -rx "database show cidname"
Because that method is not easy to maintain I would like to ask if it is possible to sync my Google contacts to that database. Then I could set a cron job to run 5 times a day.

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Re: [asterisk-users] asterisk email to fax

[tux john]

At the moment it receives fax and sends it as pdf to my email.

i have followed this tutorial http://the-asterisk-book.com/1.6/faxserver-mit-iaxmodem-und-hylafax.html

i have setup postfix to send using my gmail account.

how do i do the reverse to send from my email?

 

Sent: Thursday, July 02, 2015 at 3:02 PM
From: jg 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: Re: [asterisk-users] asterisk email to fax



 




being honest with i have been lost on what to do.

all i want is sent from my email a pdf file and then the server will sent it as fax.

what settings do i have to do regarding emailing to the server? what other settings do i have to do?

is there a guide on that?

 

Sent: Friday, June 26, 2015 at 7:28 PM
From: "Tiago Geada" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: Re: [asterisk-users] asterisk email to fax



we use a PHP web page, that takes a few formats, PDF being the most common, anc convert it to TIFF.

 

If conversion succeeds we allow to download the TIFF file as a preview. Then the user confirms and the PHP places a .call file in asterisk spool


 
On 25 June 2015 at 19:51, Ryan, Travis  wrote:




I hope his mother in law doesn’t live with him. That’s a support issue for sure.

 

 


From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Thursday, June 25, 2015 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk email to fax


 

> Since the O.P. said he's using it for his home office, I think he'll
> be able to control user expectations :-)


I provide tech support to my parents on all their computers. The amount of annoyance I have dealt with in the last few months over the fact that a recipe program and various card making programs designed for Windows 3.1/95 won't run on my mom's Windows 7 64 bit computer tells me you are not as right as you think you are.











This is not a question of settings. You must decide yourself what to do and there are various options. Asterisk is only responsible for the transport.

You need to look at the following tasks:
- get the file or files into the asterisk box
- convert to a faxable tiff format
- generate a call file and put the tiff file where it belongs (but there are other methods)
- get fax report

Of course, at first you must configure Asterisk to accept facsimiles, which depends on which technologies you need. But everything is nicely documented.

jg
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Re: [asterisk-users] asterisk email to fax

[tux john]

being honest with i have been lost on what to do.

all i want is sent from my email a pdf file and then the server will sent it as fax.

what settings do i have to do regarding emailing to the server? what other settings do i have to do?

is there a guide on that?

 

Sent: Friday, June 26, 2015 at 7:28 PM
From: "Tiago Geada" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: Re: [asterisk-users] asterisk email to fax



we use a PHP web page, that takes a few formats, PDF being the most common, anc convert it to TIFF.

 

If conversion succeeds we allow to download the TIFF file as a preview. Then the user confirms and the PHP places a .call file in asterisk spool


 
On 25 June 2015 at 19:51, Ryan, Travis  wrote:




I hope his mother in law doesn’t live with him. That’s a support issue for sure.

 

 


From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Thursday, June 25, 2015 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk email to fax


 

> Since the O.P. said he's using it for his home office, I think he'll
> be able to control user expectations :-)


I provide tech support to my parents on all their computers. The amount of annoyance I have dealt with in the last few months over the fact that a recipe program and various card making programs designed for Windows 3.1/95 won't run on my mom's Windows 7 64 bit computer tells me you are not as right as you think you are.



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[asterisk-users] asterisk email to fax

[tux john]

hello everyone.

i am using asterisk 11.16 in my home office and i am using fax to email with it. i am quite happy with the way it works, no problems at all. when a fax arrives in a particular DID then the system sends it with mailutils to my email address as pdf. there is not any email address setup in the system, simply mailutils.

 

i would like to add email to fax functionality to the system. could someone point me to the right direction to see how please?

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Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

[tux john]

a similar setup with 3 xspa3102 are working in a law office for 2 years non stop. they are very reliable, but a pain to configure them.

regarding freepbx is very helpful if you do not want to have yourself involved with pure asterisk.

 

Sent: Wednesday, June 17, 2015 at 4:12 PM
From: "Michelle Dupuis" 
To: "Asterisk Users List" 
Subject: Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

I think you are mixing up answers and general advice. FreePBX was intended to get you over the dialplan creation hurdle (the biggest challenge for people new to Asterisk).

In regards to the LinkSys they are compatible and you do find them in enterprises, but admins are trying to get rid of adapters/converters so if possible you may wish to invest in SIP devices directly instead of an adapter.

-M-


From: asterisk-users-boun...@lists.digium.com  on behalf of lu...@sulweb.org 
Sent: Wednesday, June 17, 2015 9:07 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

Lukasz Sokol wrote:
> but have you considered a web-managed config-builder such as FreePBX?
> Instead of building your dialplan from scratch ?

I've never used FreePBX, but, after having looked at its website, I
think I have a general understanding of what it can do. What I don't
understand is how FreePBX answers my question about the Linksys SPA3102
being good for a mission critical solution or not.



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Re: [asterisk-users] small homebrew pbx

[tux john]

Good day,

 

i am using a raspberry pi 2, where i am running asterisk 1.8. it runs really well.

to interface the 2 PSTN lines i am using 2x linksys spa3102

costs:

-raspberry with case and power supply 50€

-spa 3102 25€ each.

total=100€

 

also i do have a couple of sip trunks with betamax.

the PBX serves 10 users, i have seen 7 concurrent calls, at g711a.

 

it is really cool machine.

 

 

Sent: Monday, June 15, 2015 at 8:58 PM
From: "John Novack" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: Re: [asterisk-users] small homebrew pbx


 
James Cass wrote:


I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install.  Very low power draw as well.

 




James Cass

jcas...@gmail.com







The JS-200 runs a very old ( 1.4 ) version of Asterisk

I have set up more than 30 nodes using the HP thin clients, many using the available cheap T5720 units. Install the latest AstLinux in the flash, and follow the advice for a PSTN provider. I prefer voip.ms here in the US, and they also will deliver via IAX, which I prefer as SIP has so many hacking attempts I just don't want to deal with it.
AstLinux in our private peer to peer network, along with many also having a PSTN connection, is easy to set up, easy to support remotely, and with a flash based system very reliable.
also Astlinux has a built in facility for an in place upgrade. It also doesn't have the PITA configuration of a PIAF. Standard Asterisk conf files are used
The HP 5720's also have a 120-240 volt power supply, so it should work almost worldwide
Somewhat larger than a Pi, but in a decent case that could easily be mounted  on a wall somewhere and connected to the LAN

Other newer units with multiple NIC ports and AstLinux can also be your router /firewall

Unless one is running a 100 seat call center, no need for one of those huge juice hogs anymore.

John Novack
 
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Dog is my Co-pilot
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Re: [asterisk-users] asterisk & google contacts

[tux john]

I am afraid i do not know how to write that.
 

Sent: Thursday, June 11, 2015 at 2:05 PM
From: "A J Stiles" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Subject: Re: [asterisk-users] asterisk & google contacts

On Thursday 11 Jun 2015, tux john wrote:
> Hello everyone. i am running an asterisk server and i would like to have
> the contacts from google. so every inbound call with fetch the caller ID
> from google contacts and present it to my screen.

This is really three problems, as follows:

(1) Accessing the Google Contacts API to retrieve someone's details based on
their phone number.
(2) Passing the incoming caller's number to an AGI script.
(3) Displaying the details retrieved from Google on your screen.


Presuming you already know how to write a program to look up a Google
contact's details from their phone number, you just need to turn that into an
AGI script. Then, in your dialplan, pass the incoming number to that script.

If you want Asterisk itself to have anything to do with the Google data, you
will have to return them by setting channel variables within the script. Or
if you are going to use some external means to pass the data to the user, then
you can have your script fork itself, detach and return straight away.


--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] asterisk & google contacts

[tux john]

Hello everyone. i am running an asterisk server and i would like to have the contacts from google.

so every inbound call with fetch the caller ID from google contacts and present it to my screen.

could someone help me on that please?

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[asterisk-users] asterisk-gui +spa3102

[tux john]

hello everyone,

i am using asterisk 11.6 with asterisk-gui and i am stuck in setting up the 
linksys spa3102 and the sip trunk for it.
may i have some help please?

friendly,
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