Re: [asterisk-users] asterisk-users Digest, Vol 109, Issue 30

[Satish Barot]

On Fri, Aug 30, 2013 at 11:44 AM, CDR  wrote:

>
> I am stumped
> In features.conf,I programmed this
>
> [applicationmap]
> Answer0 => 0,self/both,Macro,nway_start
>
> But do I pass an argument or parameter to my macro? I tried
> Answer0 => 0,self/both,Macro,nway_start^0
> Answer0 => 0,self/both,Macro,nway_start,0
>
> but the usuar variable ${ARG1} is empty in my dialplan.
> The issue is that my macro needs to know what key was pressed.
>
>
nway_start macro will only get executed when 0 is pressed and not on any
other key press. When in macro nway_start, You can safely assume that only
0 is pressed.

--Satish Barot
Ahmedabad, India.
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Re: [asterisk-users] asterisk-users Digest, Vol 109, Issue 30

[CDR]

I am stumped
In features.conf,I programmed this

[applicationmap]
Answer0 => 0,self/both,Macro,nway_start

But do I pass an argument or parameter to my macro? I tried
Answer0 => 0,self/both,Macro,nway_start^0
Answer0 => 0,self/both,Macro,nway_start,0

but the usuar variable ${ARG1} is empty in my dialplan.
The issue is that my macro needs to know what key was pressed.
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Re: [asterisk-users] Send Fax from Asterisk

[Tahir Almas]

it support email to fax and fax to email, you are looking for

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT


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On Wed, Aug 15, 2012 at 11:35 PM, Ahmed Munir wrote:

> Thanks for sharing the link. Actually I'm looking for a different approach
> without installing/using third party i.e. a user sends an email to Asterisk
> (which is also running mail service), as Asterisk receives the mail where
> the mail contains attachment and subject contains destination  number,
> Asterisk will download the file and capture the number and later send fax
> to destination number just like '.call' file.
>
> Does anyone worked on this scenario? If yes/no, please let me know at
> earliest.
>
>
>
>
> please check it. might be it will help
>>
>> http://ictfax.org/content/installation-guide
>>
>> On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir > >wrote:
>>
>> > Hi,
>> >
>> > I would like to know, anyone who worked in Email to Fax scenario? If so
>> > please share the idea for implementing it.
>> >
>> > As on other hand I configured Asterisk  for inbound Fax which is working
>> > good i.e. later forward the fax via email but don't know how can I
>> > implement for outbound fax in this case.
>> >
>> > Please advice.
>> >
>> > --
>> > Regards,
>> >
>> > Ahmed Munir Chohan
>> >
>> >
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9718500594
>> Asterisk Developer
>> E-mail-: virbh...@gmail.com
>> Skype id:- virbhati2
>> New Delhi(India)
>> [image: View my profile on
>> LinkedIn]
>>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
>
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Re: [asterisk-users] Continue script on hangup

[Steve Edwards]

On Thu, 29 Aug 2013, John Doe wrote:

Is it possible to execute lines in a script when the caller disconnects? 
I.e. to check when the last person leaves a conference room so you know 
it is safe to move the recording file.


Your question is a little bit vague, and a few details like what version 
of Asterisk and what language you are fluent in would help.


If you are asking if you can execute an AGI in the 'h' extension, yes.

If you are asking if you can continue executing an AGI when a caller hangs 
up, yes. Asterisk will deliver a SIGHUP. If you have established a signal 
handler (man sigaction or man signal) to 'catch' or 'trap' the signal, you 
can do as you please.


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

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[asterisk-users] Continue script on hangup

[John Doe]

Is it possible to execute lines in a script when the caller disconnects?
I.e. to check when the last person leaves a conference room so you know it
is safe to move the recording file.
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Re: [asterisk-users] ReceiveFAX problem

[Markus]

Hi Matthias,

Am 29.08.2013 23:17, schrieb Matthias Leopold:

today i upgraded from asterisk 1.4.21 to 1.6.2.9 (i know this release is
not supported anymore, please don't tell me to upgrade).

unfortunately now i can't use the rxfax() application anymore.


I don't know about rxfax/ReceiveFAX, but I'm using iaxmodem and HylaFAX 
to receive faxes and send them to different recipients depending on the 
extension that the fax was received on.


I wrote a small how-to which you can find here: 
http://truemetal.org/universe/blog/2013/03/receiving-faxes-with-asterisk-iaxmodem-and-hylafax-with-dynamic-email-recipients/


HTH
Markus


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[asterisk-users] ReceiveFAX problem

[Matthias Leopold]

hi,

today i upgraded from asterisk 1.4.21 to 1.6.2.9 (i know this release is 
not supported anymore, please don't tell me to upgrade).


unfortunately now i can't use the rxfax() application anymore. i tried 
to use ReceiveFAX() the way i used to use rxfax() (multiple dedicated 
fax extensions getting their faxes from isdn via zaptel/dahdi), but this 
doesn't work.


either fax receiving "works" (tiff file there), but the result isn't 
mailed like described here


http://forums.asterisk.org/viewtopic.php?p=151415

or in a slightly different setup i get "Fax detected, but no fax extension".

i need multiple fax extensions. how do i do this? separate contexts for 
each fax extension with a "fax extension"? i didn't manage to do this.


i know i didn't post any config and my description is rather crude, but 
maybe someone understands my problem and can point me in the right 
direction.


thx
matthias


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Re: [asterisk-users] How to reply with 480 Call-limit to incoming SIP call ?

[Gareth Blades]

On 29/08/13 14:42, Olivier wrote:

Thanks for your very helpful reply.

1.My system prints out:
CLI> core show application Hangup

  -= Info about application 'Hangup' =-

[Synopsis]
Hang up the calling channel.

[Description]
This application will hang up the calling channel.

[Syntax]
Hangup([causecode])

[Arguments]
causecode
If a  is given the channel's hangup cause will be set
to the given value.

[See Also]
Answer(), Busy(), Congestion()

How could we improve this Arguments section so that other Asterisk 
admins can find available  values ?




Have a look in the source code in channels/chan_sip.c and you will see :-

const char *hangup_cause2sip(int cause)
{
switch (cause) {
case AST_CAUSE_UNALLOCATED: /* 1 */
case AST_CAUSE_NO_ROUTE_DESTINATION:/* 3 IAX2: 
Can't find extension in context */

case AST_CAUSE_NO_ROUTE_TRANSIT_NET:/* 2 */
return "404 Not Found";
case AST_CAUSE_CONGESTION:  /* 34 */
case AST_CAUSE_SWITCH_CONGESTION:   /* 42 */
return "503 Service Unavailable";
case AST_CAUSE_NO_USER_RESPONSE:/* 18 */
return "408 Request Timeout";
case AST_CAUSE_NO_ANSWER:   /* 19 */
case AST_CAUSE_UNREGISTERED:/* 20 */
return "480 Temporarily unavailable";
case AST_CAUSE_CALL_REJECTED:   /* 21 */
return "403 Forbidden";
case AST_CAUSE_NUMBER_CHANGED:  /* 22 */
return "410 Gone";
case AST_CAUSE_NORMAL_UNSPECIFIED:  /* 31 */
return "480 Temporarily unavailable";
case AST_CAUSE_INVALID_NUMBER_FORMAT:
return "484 Address incomplete";
case AST_CAUSE_USER_BUSY:
return "486 Busy here";
case AST_CAUSE_FAILURE:
return "500 Server internal failure";
case AST_CAUSE_FACILITY_REJECTED:   /* 29 */
return "501 Not Implemented";
case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
return "503 Service Unavailable";
/* Used in chan_iax2 */
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
return "502 Bad Gateway";
case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:   /* 
Can't find codec to connect to host */

return "488 Not Acceptable Here";
case AST_CAUSE_INTERWORKING:/* Unspecified 
Interworking issues */

return "500 Network error";

case AST_CAUSE_NOTDEFINED:
default:
ast_debug(1, "AST hangup cause %d (no match 
found in SIP)\n", cause);

return NULL;
}

For any given hangup cause you can change the sip response there. For a 
list of the hangup numbers and the internal variable name look in 
include/asterisk/causes.h


So if you change chan_sip.c and add the following just before the 
'AST_CAUSE_NOTDEFINED' line and recompile and reinstall you should in 
theory be able to do a Hangup(44) to achieve what you want.


case AST_CAUSE_REQUESTED_CHAN_UNAVAIL:/* 44 */
return "480 Temporarily Unavailable (Call limit)";

Thats only in theory. I havent tested it myself and I am not an asterisk 
developer.


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Re: [asterisk-users] How to reply with 480 Call-limit to incoming SIP call ?

[Olivier]

Thanks for your very helpful reply.

1.My system prints out:
CLI> core show application Hangup

  -= Info about application 'Hangup' =-

[Synopsis]
Hang up the calling channel.

[Description]
This application will hang up the calling channel.

[Syntax]
Hangup([causecode])

[Arguments]
causecode
If a  is given the channel's hangup cause will be set
to the given value.

[See Also]
Answer(), Busy(), Congestion()

How could we improve this Arguments section so that other Asterisk admins
can find available  values ?




2013/8/20 Rusty Newton 

> On Fri, Aug 16, 2013 at 2:22 AM, Olivier  wrote:
> > Hi,
> >
> > After Googling, I found information on how you can read the status of an
> > outgoing call but I didn't find anything on tunning reply to incoming
> calls.
> >
> > My question is :
> >
> > I've got a system receiving SIP calls from different callers.
> > I would like to end some calls with a "480 Temporarily Unavailable (Call
> > limit)" reply
> > Is it possible ?
>
> Should be!
>
> Take a look at
> https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
>
> Then try using the Hangup() application and passing the appropriately
> mapped cause code. Sounds like 19 in this case.
>
> So using Hangup(19) in your dialplan may send out the 480 at least.
>
> --
> Rusty Newton
> Digium, Inc. | Community Support Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] Kepress while on Queue

[Lenz Emilitri]

Yes it will work. One interesting option here is adding to the MOH an
invitation to exit and leave your number and the CC will call you back.
Helps you smooth the load during peak times, reduces staff and everyone
wins :)
l.



2013/8/27 Gopalakrishnan N 

> Hi,
>
> Will Keypress option will work when am in the queue and hearing MoH?
>
> Lets say a caller is waiting in queue and while he is hearing MoH, can he
> key in some DTMF and go to some other queue? is that possible?
>
> Regards
>
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Re: [asterisk-users] Need input on scalable system design...

[Lenz Emilitri]

Hi Greg,
I am aware of a couple of solutions that come prepackaged and offer
distributed queues for Asterisk. One of them, that seems to work well and
reliably, is the one from Raynet. I am sure there are more. On the other
side, I have seen a number of in-house solutions where you basically have a
daemon polling queues statuses and redirecting calls based on the relative
wait times. Rough but effective, and can be deployed easily.

About recordings, my suggestion would be to use something to offload them
right from the servers, like Oreka. Have a number of large clients using it
and they are quite happy (plus, the guys supporting it are superb).

Just my two cents,
l.




2013/8/27 Gregory Malsack 

>   Hey All,
>
> Growing call center. Currently at about 200 call center staff, running
> about 1000 calls per hour. Gearing up to double that. Not too sure that a
> single server will support that growth. So, I'm trying to come up with ways
> to scale the system and still maintain a simplistic design. So I'd like to
> bounce some ideas around.
>
> Currently I am running on a Dell 1950, dual quad core 2.33ghz xeons, with
> 16gb ram, and 2 tce400p cards. This server is managing the full load of the
> company. We are recording all calls, running ivr, queues, cdr, cel, and web
> for reporting. I currently have another 1950 of the exact same
> specifications as a cold spare.
>
> Here's where you can see drawings of my current connectivity and an
> optional connectivity I'm contemplating...
>
> http://www.paydaysupportcenter.com/current.pdf
> http://www.paydaysupportcenter.com/option.pdf
>
> As you can see I currently have a separate sql server and a separate
> storage server for the call recordings. This is all working fine.
>
> However, I'm thinking for scalability I should be looking to migrate to a
> configuration similar to the one in option.pdf. Where I have a VOIP gateway
> server that simply relays traffic and possibly can do some load balancing
> or intellegent routing. But nothing more then that, and possibly a second
> one of these online as a hot failover.
>
> Then have separate sql, storage, (i forgot it in the pic) web, and
> asterisk servers behind that on separate dedicated network. Here's my
> dilemma though, how do I balance the load across multiple machines for
> scalability...
>
> Since 95% of our calls come into queues, I need to be able to maintain
> queue stats and presence across all of the servers. Thus far, I've got
> everything except the extensions.conf file into the mysql database. I
> thought about setting up 2 servers, 1 for sales, and 1 for customer
> service, then possibly break out each call queue to it's own server as
> things grow. Just not sure if that's the right way to go.
>
> Then regarding extensions.conf, I've read that it too can be placed in the
> sql database and accessed via switch. however it's resource intense, so now
> I'm thinking of maybe putting that file on the nfs server for all of the
> boxes to read from.
>
> As for the design of that file, I was kind of thinking of a modular design
> within the file using various goto's and gosubs. Our business model is
> based on affiliates and corporate marketing, so we have a ton of did's that
> follow the same call flow with minor modifications in some variables, as
> well as variations in call flow, and hours of operation. Thus the modular
> design of the call flow. Then the primary inbound context would simply be a
> list of did's pointing to a goto with a list of the variations and
> variables for the did.
>
> Ok, now that I've melted your brains thoughts?
>
> Thanks all in advance for the discussion...
> Greg
>
> --
> Greg
>
>
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Re: [asterisk-users] G729 Passthrough How To

[Nick Cameo]

You ok sir? Are you going to make it?

N.

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Re: [asterisk-users] channel driver development in Ireland

[Gustavo Meira]

I've sent them an email. Thank you, Laszlo.


2013/8/27 Laszlo 

> Hi,
>
> Maybe you can get in touch with them:
>
> http://www.innovate.ie
>
> - Laszlo
>
>
>
>
> 2013/8/27 Gustavo Meira 
>
>>  Hi, Guys.
>>
>> I wanted to know from you, if you know: 1) any companies in Ireland that
>> develop software and hardware for Asterisk integration with legacy
>> interfaces (E1, T1, FXO, FXS...) . 2) any companies in Ireland that work
>> with Asterisk development (modules, channel drivers...).
>>
>> I work with channel driver development and I'm moving to Ireland in
>> November, so I wanted to see how these things are going there.
>>
>> Thanks in advance.
>> --
>> Gustavo Roberto Nardon Meira
>>
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>
>
>
> --
>
> --
> Kind regards,
> Laszlo Bekesi
> http://voipfreak.net
>
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-- 
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Re: [asterisk-users] Installing asterisk and dahdi on ubuntu

[Duncan Turnbull]

On 29/08/2013, at 10:02 PM, Thorsten Göllner  wrote:

> Permissions: take a look at "/etc/udev/rules.d/dahdi.rules". Last line. OWNER 
> and GROUP should be the same as the user running the asterisk process (root 
> or asterisk?).
> 
> Am 29.08.2013 11:47, schrieb bilal ghayyad:
>> Hello;
>> 
>> I am installing asterisk and dahdi on ubuntu "and I used my username bghayad 
>> to login for ubuntu and do the installation, actually I feel my problem is 
>> related to the username and permission but I am not able how to fix it", I 
>> am facing now mainly the following two problems:
>> 
>> The first one, asterisk is not starting automatically although I did sudo 
>> make config (for asterisk and dahdi) and the asterisk and dahdi scripts have 
>> been created under /etc/init.d/
>> 
>> The second problem, I started asterisk using asterisk -cvvv and from the 
>> CLI, I tried dahdi show version and dahdi show status, I am getting the 
>> following results:
>> 

Also did you start dahdi? /etc/init.d/dahdi start

This happens when its not running

In asterisk you can module load chan_dahdi but if dahdi isn't running then it 
won't help


>> *CLI> dahdi show status
>> No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied
>> Command 'dahdi show status ' failed.
>> 
>> *CLI> dahdi show version
>> Failed to open control file to get version.
>> 
>> 
>> Below is my ubuntu information:
>> 
>> bghayad@Bilal:/usr/sbin$ lsb_release -a
>> No LSB modules are available.
>> Distributor ID: Ubuntu
>> Description:Ubuntu 12.04.1 LTS
>> Release:12.04
>> Codename:   precise
>> 
> 
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Re: [asterisk-users] Installing asterisk and dahdi on ubuntu

[Thorsten Göllner]

Permissions: take a look at "/etc/udev/rules.d/dahdi.rules". Last line. 
OWNER and GROUP should be the same as the user running the asterisk 
process (root or asterisk?).


Am 29.08.2013 11:47, schrieb bilal ghayyad:

Hello;

I am installing asterisk and dahdi on ubuntu "and I used my username 
bghayad to login for ubuntu and do the installation, actually I feel 
my problem is related to the username and permission but I am not able 
how to fix it", I am facing now mainly the following two problems:


The first one, asterisk is not starting automatically although I did 
sudo make config (for asterisk and dahdi) and the asterisk and dahdi 
scripts have been created under /etc/init.d/


The second problem, I started asterisk using asterisk -cvvv and from 
the CLI, I tried dahdi show version and dahdi show status, I am 
getting the following results:


*CLI> dahdi show status
No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied
Command 'dahdi show status ' failed.

*CLI> dahdi show version
Failed to open control file to get version.


Below is my ubuntu information:

bghayad@Bilal:/usr/sbin$ lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:Ubuntu 12.04.1 LTS
Release:12.04
Codename:   precise



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Re: [asterisk-users] Installing asterisk and dahdi on ubuntu

[A J Stiles]

On Thursday 29 August 2013, bilal ghayyad wrote:
> Hello;
> 
> I am installing asterisk and dahdi on ubuntu "and I used my username
> bghayad to login for ubuntu and do the installation, actually I feel my
> problem is related to the username and permission but I am not able how to
> fix it", I am facing now mainly the following two problems:

Just because the initscripts exist in .etc/init.d, does not mean that they are 
symbolically linked from /etc/rc[0-6].d.  I think Ubuntu uses "update-rc.d" to 
do this, but you will have to refer to your documentation.

To deal with the permissions thing, try
$ sudo su -
which will give you a normal root prompt.

-- 
AJS

Answers come *after* questions.

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[asterisk-users] Installing asterisk and dahdi on ubuntu

[bilal ghayyad]

Hello;

I am installing asterisk and dahdi on ubuntu "and I used my username bghayad to 
login for ubuntu and do the installation, actually I feel my problem is related 
to the username and permission but I am not able how to fix it", I am facing 
now mainly the following two problems:

The first one, asterisk is not starting automatically although I did sudo make 
config (for asterisk and dahdi) and the asterisk and dahdi scripts have been 
created under /etc/init.d/

The second problem, I started asterisk using asterisk -cvvv and from the CLI, I 
tried dahdi show version and dahdi show status, I am getting the following 
results:

*CLI> dahdi show status
No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied
Command 'dahdi show status ' failed.

*CLI> dahdi show version
Failed to open control file to get version.


Below is my ubuntu information:

bghayad@Bilal:/usr/sbin$ lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:    Ubuntu 12.04.1 LTS
Release:        12.04
Codename:       precise

Regards
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Re: [asterisk-users] Asterisk 1.8.15-cert3, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.5.1 Now Available (Security Release)

[Gareth Blades]

On 28/08/13 19:34, Rusty Newton wrote:

On Wed, Aug 28, 2013 at 11:26 AM, Gareth Blades
  wrote:

On 27/08/13 19:20, Asterisk Development Team wrote:

The Asterisk Development Team has announced security releases for
Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available
security releases
are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3,
10.12.3-digiumphones,
and 11.5.1.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

http://www.asterisk.org/downloads/asterisk/all-asterisk-versions sais :-
Latest Version - 11.2-cert2

However if you download it its the old cert1 version.

Going direct to :-
http://downloads.asterisk.org/pub/telephony/certified-asterisk/
there is no cert2 version there yet.

It looks good now. Perhaps you were looking at cached versions of
those pages?  Were you eventually able to get the cert2 versions?


I can see it there now fine thanks. Its strange as I also downloaded the 
changelog.current file which I hadnt downloaded before and even that was 
the old version. The cert1 tar.gz isnt there any more and as I did end 
up downloading that version I think it just took a while until the new 
files were replicated onto the server.


Thanks



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