Re: [asterisk-users] dahdi on systemd (CentOS 7)

[Tzafrir Cohen]

On Tue, Feb 02, 2016 at 10:07:42AM -0500, Jerry Geis wrote:
> It doesnt appear dahdi is starting up under systemd and CentOS 7.2
> 
> What should I look for?
> 
> find /etc/systemd | grep -i dahdi
> 
> find nothing.

/etc/systemd is for services installed by the system administrator.

You should have also looked under /usr/lib/systemd .

Alternatively:

  systemctl status dah # completes to:
  systemctl status dahdi.service

Which will shows me that dahdi.service was generated from
/etc/init.d/dahdi .

Or:

   systemctl list-units | grep dahdi 

> 
> How does dahdi startup under systemd ?

Right now with the same init.d script. That said, I'd like to avoid
using it. If you switch to automatic span assignment
(http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments -
auto_assign_spans=0), you don't really need the DAHDI init script. In
fact, using it may become confusing, as under some circumstances the
dahdi "service" may be in the wrong state for you (and unloading modules
at poweroff is pointless).

Thus, in the spirit of the parallel boot of systemd, my general
recommendation is not to have any dahdi service, and just let spans load
and get initialized separately. This also means you no longer need to
start DAHDI before Asterisk.

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Re: [asterisk-users] Dial command: channel type detection

[jg]

Some of my users connect to my asterisk box using SIP, other using iax
(in users.conf, I set "hasiax=yes" for those users).

How do I detect which protocol some user is using ? I cannot find any
variable which contains that information.

Reason is: I need this information for the Dial() command to work with
all my users, as the protocol is needed when using this command.

Why can't you evaluate the CHANNEL variable with something like 
Set(TECHNOLOGY=${CUT(CHANNEL,/,1)})? One could also initially use a special context for IAX 
channels and set a variable. It depends on what you want to do.


jg

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Re: [asterisk-users] Compile error with libpri 1.4.15

[Tzafrir Cohen]

On Tue, Feb 02, 2016 at 11:57:08AM -0500, Jerry Geis wrote:

> make[4]: Entering directory
> `/home/silentm/LayeredSolutions/digium/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp'
> /usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or
> directory
> /usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or
> directory
> /usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory

Those should have been generated by pod2man. Look for '%.8: %' in
xpp/Makefile.am .

So it seems that a check for pod2man needs to be added to the configure
script. For now, generate a dummy pod2man binary somewhere in your path.
Or just install perl-doc or whatever it is called.

For instance:

  ln -s /bun/true /usr/local/bin/pod2man # Results in empty man pages

I stress that this is a TEMPORARY WORKAROUND until the configure script
is adapted.

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Re: [asterisk-users] How to deal with error messages passed as Early Media

[Duncan]

On 04/02/16 06:00, Scott Griepentrog wrote:
For calls that fail, even where early media is played, the call should 
terminate with a 4xx or 5xx SIP response which to a certain degree 
correlates to the nature of the actual failure.  The SIP error code is 
delayed until the media playback completes, but should be no different 
whether or not early media is used (for the same actual failure).


Early media is simply an audio stream for human consumption to explain 
the failure.  There should be no need to attempt to recognize it, 
unless your ITSP is not terminating the call correctly.


I recently ran some analysis of early media messages found in cellular 
networks by recording the calls and running it through CMU-Sphinx. There 
are only a few types of early media messages per network, to cover a 
raft of failures.


But I always got a SIP error code back as well, the early media I found 
tended to play for upto 20 secs then drop the call, then you get the 
error code. It might not be a very descriptive error but its still an 
error code, and the early media audio message is not always very 
distinct either. In the GSM networks the GSM failure code is more useful 
but still seems somewhat randomly assigned by the provider, even 
including the odd temporary failures.


You are not really worried about what the failure reason is, its the 
caller who needs to decide - did I misdial, is the number really 
disconnected, is their phone out of coverage etc, you just need to try 
your next available network, if the caller hasn't already hung up after 
hearing the message


You could possibly examine the audio before you get an answer but then 
you might get caught by some other system or PBX playing early media 
before answer that isn't actually a failure.


If your ITSP is not giving you an error code then you have an issue.

Cheers Duncan



On Wed, Feb 3, 2016 at 8:41 AM, Olivier > wrote:


Hello,

I'm trunking with an ITSP that, when treating an outbound to an
unknown destination, either:
- send a SIP error code (I can't be more explicit, at the moment),
- or cast a pre-recorded audio message using Early Media.

At the same time, I'm also trunking with Contact Center solution
which doesn't support Early Media.


Beside asking my ITSP to treat calls consistently or ask  Contact
Centerto support Early Media, is there a way to configure Asterisk
to unify both above error treaments into a single one ?

How can I best deal with error messages passed as Early Media.

Best regards

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[asterisk-users] What is SIP Early Media useful for ?

[Olivier]

Hello,

Could you help me to summarize what is SIP Early Media useful for ?

I was thinking of:
- Passing error messages to caller,
- Custom ringing tones to caller.

Did I miss something ?

Best regards
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Re: [asterisk-users] How to deal with error messages passed as Early Media

[Olivier]

2016-02-03 15:59 GMT+01:00 Steve Howes :

>
>
> On 03/02/16 14:41, Olivier wrote:
>
>> How can I best deal with error messages passed as Early Media.
>>
> Tell the ITSP to give you proper signaling, if they wont then get a new
> ITSP. I suspect if they can't handle this correctly, there will be a lot
> more they're doing wrong as well. Long term you'll save yourself a whole
> lot of bother.
>
>
Yes but I'm afraid that, in this industry, the rule is to pass anything
received to the other party.


Steve
>
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Re: [asterisk-users] How to deal with error messages passed as Early Media

[Steve Howes]

 On 03/02/16 15:29, Olivier wrote:
2016-02-03 15:59 GMT+01:00 Steve Howes >:




On 03/02/16 14:41, Olivier wrote:

How can I best deal with error messages passed as Early Media.

Tell the ITSP to give you proper signaling, if they wont then get
a new ITSP. I suspect if they can't handle this correctly, there
will be a lot more they're doing wrong as well. Long term you'll
save yourself a whole lot of bother.


Yes but I'm afraid that, in this industry, the rule is to pass 
anything received to the other party.
In that case I wish you the best of luck. You can't process audio and 
turn it into a proper signal. If they don't send a SIP/ISDN signal then 
you're stuffed.


I still maintain the best way is to get the right thing sent to you in 
the first place - it's a basic interop requirement that data is 
consistent (even if it's not exactly the format what you want)


Steve
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Re: [asterisk-users] How to deal with error messages passed as Early Media

[Steve Howes]

On 03/02/16 14:41, Olivier wrote:

How can I best deal with error messages passed as Early Media.
Tell the ITSP to give you proper signaling, if they wont then get a new 
ITSP. I suspect if they can't handle this correctly, there will be a lot 
more they're doing wrong as well. Long term you'll save yourself a whole 
lot of bother.


Steve

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[asterisk-users] How to deal with error messages passed as Early Media

[Olivier]

Hello,

I'm trunking with an ITSP that, when treating an outbound to an unknown
destination, either:
- send a SIP error code (I can't be more explicit, at the moment),
- or cast a pre-recorded audio message using Early Media.

At the same time, I'm also trunking with Contact Center solution which
doesn't support Early Media.


Beside asking my ITSP to treat calls consistently or ask  Contact Centerto
support Early Media, is there a way to configure Asterisk to unify both
above error treaments into a single one ?

How can I best deal with error messages passed as Early Media.

Best regards
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[asterisk-users] Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)

[Asterisk Development Team]

The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.

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

The release of these versions resolves the following security vulnerabilities:

* AST-2016-001: BEAST vulnerability in HTTP server

  The Asterisk HTTP server currently has a default configuration which allows
  the BEAST vulnerability to be exploited if the TLS functionality is enabled.
  This can allow a man-in-the-middle attack to decrypt data passing through it.

* AST-2016-002: File descriptor exhaustion in chan_sip

  Setting the sip.conf timert1 value to a value higher than 1245 can cause an
  integer overflow and result in large retransmit timeout times. These large
  timeout values hold system file descriptors hostage and can cause the system
  to run out of file descriptors.

* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.

  If no UDPTL packets are lost there is no problem. However, a lost packet
  causes Asterisk to use the available error correcting redundancy packets. If
  those redundancy packets have zero length then Asterisk uses an uninitialized
  buffer pointer and length value which can cause invalid memory accesses later
  when the packet is copied.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf

Thank you for your continued support of Asterisk!


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[asterisk-users] AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2016-003

 ProductAsterisk  
 SummaryRemote crash vulnerability when receiving UDPTL FAX   
data. 
Nature of Advisory  Denial of Service 
  SusceptibilityRemote Authenticated Sessions 
 Severity   Minor 
  Exploits KnownYes   
   Reported On  December 2, 2015  
   Reported By  Walter Dokes, Torrey Searle   
Posted On   February 3, 2016  
 Last Updated OnFebruary 3, 2016  
 Advisory Contact   Richard Mudgett   
 CVE Name   Pending   

Description  If no UDPTL packets are lost there is no problem. However,   
 a lost packet causes Asterisk to use the available error 
 correcting redundancy packets. If those redundancy packets   
 have zero length then Asterisk uses an uninitialized buffer  
 pointer and length value which can cause invalid memory  
 accesses later when the packet is copied.

Resolution  Upgrade to a released version with the fix incorporated or
apply patch.  

   Affected Versions   
 Product   Release  
   Series   
  Asterisk Open Source  1.8.x   All versions  
  Asterisk Open Source  11.xAll versions  
  Asterisk Open Source  12.xAll versions  
  Asterisk Open Source  13.xAll versions  
   Certified Asterisk  1.8.28   All versions  
   Certified Asterisk   11.6All versions  
   Certified Asterisk   13.1All versions  

  Corrected In
  Product  Release
Asterisk Open Source   11.21.1, 13.7.1
 Certified Asterisk11.6-cert12, 13.1-cert3

  Patches  
 SVN URL   Revision 
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-1.8.28.diff 
Certified 
   Asterisk 
 
   1.8.28   
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-11.6.diff   
Certified 
   Asterisk 
 
   11.6 
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-13.1.diff   
Certified 
   Asterisk 
 
   13.1 
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-1.8.diffAsterisk 
 
   1.8  
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-11.diff Asterisk 
 
   11   
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-12.diff Asterisk 
 
   12   
 
   http://downloads.asterisk.org/pub/security/AST-2016-003-13.diff Asterisk 
 
   13   
 

Links  https://issues.asterisk.org/jira/browse/ASTERISK-25603 

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
http://downloads.digium.com/pub/security/AST-2016-003.pdf and 
http://downloads.digium.com/pub/security/AST-2016-003.html

Revision History
  Date  

[asterisk-users] AST-2016-001: BEAST vulnerability in HTTP server

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2016-001

 ProductAsterisk  
 SummaryBEAST vulnerability in HTTP server
Nature of Advisory  Unauthorized data disclosure due to   
man-in-the-middle attack  
  SusceptibilityRemote unauthenticated sessions   
 Severity   Minor 
  Exploits KnownYes   
   Reported On  04/15/15  
   Reported By  Alex A. Welzl 
Posted On   02/03/16  
 Last Updated OnFebruary 3, 2016  
 Advisory Contact   Joshua Colp  
 CVE Name   Pending   

Description  The Asterisk HTTP server currently has a default 
 configuration which allows the BEAST vulnerability to be 
 exploited if the TLS functionality is enabled. This can  
 allow a man-in-the-middle attack to decrypt data passing 
 through it.  

Resolution  Additional configuration options have been added to Asterisk  
which allow configuration of the HTTP server to not be
susceptible to the BEAST vulnerability. These include 
options to confirm the permitted ciphers, to control what 
TLS protocols are allowed, and to use server cipher   
preference order instead of client preference order. The  
default configuration has also been changed for the HTTP  
server to use a configuration which is not susceptible to 
the BEAST vulnerability.  

   Affected Versions   
 Product   Release  
   Series   
  Asterisk Open Source  1.8.x   All Versions  
  Asterisk Open Source  11.xAll Versions  
  Asterisk Open Source  12.xAll Versions  
  Asterisk Open Source  13.xAll Versions  
   Certified Asterisk  1.8.28   All Versions  
   Certified Asterisk   11.6All Versions  
   Certified Asterisk   13.1All Versions  

  Corrected In
  Product  Release
Asterisk Open Source   11.21.1, 13.7.1
 Certified Asterisk11.6-cert12, 13.1-cert3

  Patches  
 SVN URL   Revision 
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-1.8.28.diff 
Certified 
   Asterisk 
 
   1.8.28   
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-11.6.diff   
Certified 
   Asterisk 
 
   11.6 
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-13.1.diff   
Certified 
   Asterisk 
 
   13.1 
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-11.diff Asterisk 
 
   11   
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-12.diff Asterisk 
 
   12   
 
   http://downloads.asterisk.org/pub/security/AST-2016-001-13.diff Asterisk 
 
   13   
 

Links  https://issues.asterisk.org/jira/browse/ASTERISK-24972 

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  

[asterisk-users] AST-2016-002: File descriptor exhaustion in chan_sip

[Asterisk Security Team]

   Asterisk Project Security Advisory - AST-2016-002

 ProductAsterisk  
 SummaryFile descriptor exhaustion in chan_sip
Nature of Advisory  Denial of Service 
  SusceptibilityRemote Unauthenticated Sessions   
 Severity   Minor 
  Exploits KnownYes   
   Reported On  September 17, 2015
   Reported By  Alexander Traud   
Posted On   February 3, 2016  
 Last Updated OnFebruary 3, 2016  
 Advisory Contact   Richard Mudgett   
 CVE Name   Pending   

Description  Setting the sip.conf timert1 value to a value higher than
 1245 can cause an integer overflow and result in large   
 retransmit timeout times. These large timeout values hold
 system file descriptors hostage and can cause the system to  
 run out of file descriptors. 

Resolution  Setting the sip.conf timert1 value to 1245 or lower will not  
exhibit the vulnerability. The default timert1 value is 500.  
Asterisk has been patched to detect the integer overflow and  
calculate the previous retransmission timer value.

   Affected Versions   
 Product   Release  
   Series   
  Asterisk Open Source  1.8.x   All versions  
  Asterisk Open Source  11.xAll versions  
  Asterisk Open Source  12.xAll versions  
  Asterisk Open Source  13.xAll versions  
   Certified Asterisk  1.8.28   All versions  
   Certified Asterisk   11.6All versions  
   Certified Asterisk   13.1All versions  

  Corrected In
  Product  Release
Asterisk Open Source   11.21.1, 13.7.1
 Certified Asterisk11.6-cert12, 13.1-cert3

  Patches  
 SVN URL   Revision 
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-1.8.28.diff 
Certified 
   Asterisk 
 
   1.8.28   
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-11.6.diff   
Certified 
   Asterisk 
 
   11.6 
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-13.1.diff   
Certified 
   Asterisk 
 
   13.1 
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-1.8.diffAsterisk 
 
   1.8  
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-11.diff Asterisk 
 
   11   
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-12.diff Asterisk 
 
   12   
 
   http://downloads.asterisk.org/pub/security/AST-2016-002-13.diff Asterisk 
 
   13   
 

Links  https://issues.asterisk.org/jira/browse/ASTERISK-25397 

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
http://downloads.digium.com/pub/security/AST-2016-002.pdf and 
http://downloads.digium.com/pub/security/AST-2016-002.html

Revision History
   Date 

Re: [asterisk-users] How to deal with error messages passed as Early Media

[Scott Griepentrog]

For calls that fail, even where early media is played, the call should
terminate with a 4xx or 5xx SIP response which to a certain degree
correlates to the nature of the actual failure.  The SIP error code is
delayed until the media playback completes, but should be no different
whether or not early media is used (for the same actual failure).

Early media is simply an audio stream for human consumption to explain the
failure.  There should be no need to attempt to recognize it, unless your
ITSP is not terminating the call correctly.



On Wed, Feb 3, 2016 at 8:41 AM, Olivier  wrote:

> Hello,
>
> I'm trunking with an ITSP that, when treating an outbound to an unknown
> destination, either:
> - send a SIP error code (I can't be more explicit, at the moment),
> - or cast a pre-recorded audio message using Early Media.
>
> At the same time, I'm also trunking with Contact Center solution which
> doesn't support Early Media.
>
>
> Beside asking my ITSP to treat calls consistently or ask  Contact Centerto
> support Early Media, is there a way to configure Asterisk to unify both
> above error treaments into a single one ?
>
> How can I best deal with error messages passed as Early Media.
>
> Best 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
>



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445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Doug Lytle]

>>> On Feb 3, 2016, at 2:19 PM, Vitor Mazuco vitor.maz...@gmail.com wrote:

>>> Humm, thanks for your reply
>>> But whats is the code in parkedcalls context.
>>> Please, can you give an example?


[ramais]

include => parkedcalls


Doug

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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Ah no, I'm asking what code I put inside of parkedcalls?

This example works?

[ramais]

include => parkedcalls

[parkedcalls]
exten => 700,1,ParkedCall(701)
exten => 702,1,ParkedCall(702)
exten => 703,1,ParkedCall(703)
exten => 704,1,ParkedCall(704)

This exten works?


2016-02-03 17:27 GMT-02:00, Doug Lytle :
 On Feb 3, 2016, at 2:19 PM, Vitor Mazuco vitor.maz...@gmail.com wrote:
>
 Humm, thanks for your reply
 But whats is the code in parkedcalls context.
 Please, can you give an example?
>
>
> [ramais]
>
> include => parkedcalls
>
>
> Doug
>
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Doug Lytle]

>>> On Feb 3, 2016, at 2:32 PM, Vitor Mazuco vitor.maz...@gmail.com wrote:

>>> Ah no, I'm asking what code I put inside of parkedcalls?

Nothing,

The context parkedcalls is generated by features.conf, you just need to include 
it in your dialplan

CLI> dialplan show parkedcalls

[ Context 'parkedcalls' created by 'features' ]
  '700' =>  1. Park() [features]


Doug

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[asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Hi!

I tried to use Parking Calls

I use Asterisk 13, but I can't park any calls and it returns me

[Feb  3 16:56:11] WARNING[1693]: pbx.c:12543
ast_context_verify_includes: Context 'ramais' tries to include
nonexistent context 'parkedcalls'

What is the correct code for put in extensions.conf?

Can be this example below?

[parkedcalls]
exten => 700,1,ParkedCall(701)
exten => 702,1,ParkedCall(702)
exten => 703,1,ParkedCall(703)
exten => 704,1,ParkedCall(704)

If not, somebody knows that?

Thanks

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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Richard Mudgett]

On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco  wrote:

> Hi!
>
> I tried to use Parking Calls
>
> I use Asterisk 13, but I can't park any calls and it returns me
>
> [Feb  3 16:56:11] WARNING[1693]: pbx.c:12543
> ast_context_verify_includes: Context 'ramais' tries to include
> nonexistent context 'parkedcalls'
>

Are you loading res_parking.so?

Does your res_parking.conf define a parkext and specify the context?
Documented in configs/samples/res_parking.conf.sample:
parkext => 700  ; What extension to dial to park.
(optional; if
; specified, extensions will be created for
parkext and
; the whole range of parkpos)
context => parkedcalls  ; Which context parked calls and the
default park

Once that is configured you can include the parkedcalls context into
your ramais context.

Richard
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Humm, thanks for your reply

But whats is the code in parkedcalls context.

Please, can you give an example?

Thanks very much.


2016-02-03 17:15 GMT-02:00, Richard Mudgett :
> On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco 
> wrote:
>
>> Hi!
>>
>> I tried to use Parking Calls
>>
>> I use Asterisk 13, but I can't park any calls and it returns me
>>
>> [Feb  3 16:56:11] WARNING[1693]: pbx.c:12543
>> ast_context_verify_includes: Context 'ramais' tries to include
>> nonexistent context 'parkedcalls'
>>
>
> Are you loading res_parking.so?
>
> Does your res_parking.conf define a parkext and specify the context?
> Documented in configs/samples/res_parking.conf.sample:
> parkext => 700  ; What extension to dial to park.
> (optional; if
> ; specified, extensions will be created for
> parkext and
> ; the whole range of parkpos)
> context => parkedcalls  ; Which context parked calls and the
> default park
>
> Once that is configured you can include the parkedcalls context into
> your ramais context.
>
> Richard
>

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Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

[Sonny Rajagopalan]

Thanks to everyone for your responses. I grappled with this the past few
days, but I have to confess any amount of poking around AMI has not yielded
any result. I find the ARI based method most convenient as I am familiar
with that interface already. I did not discover this as the ARI method is
not documented under ARI information at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ARI (but instead
under a different left-hand navigation tree in your wiki). But thanks, Matt.

At any rate, I would like to setup pjsip using a mysql, rather than an
astdb based setup with uses a LOCAL db. Is there an online document for
configuring the mysql database backend for asterisk. If
res_config_mysql.conf is setup, does it mean automatically that it is used
instead of astdb sqllite? That part is not clear. Any help or documentation
is appreciated.

Thanks,
Sonny.

On Fri, Jan 29, 2016 at 9:47 AM, Matthew Jordan  wrote:

>
>
>
> On Fri, Jan 29, 2016 at 6:15 AM, Bryant Zimmerman 
> wrote:
>
>> Sonny
>>
>> We use a real-time database for adding pjsip users. If you want to do it
>> from the pjsip.conf you would have to write to the file from a script of
>> some sort and then trigger a reload.   There is a real-time implementation
>> for the extensions.conf as well. I personally use scripts for most of my
>> dialplan, but in some cases I write to files included in my dialplan from a
>> script and force a reload.
>>
>> To directly answer you question I do not believe there is an API baked
>> into asterisk to update the pjsip.conf and extensions.conf directly from
>> the dialplan.
>>
>> Thanks
>>
>> Bryant
>>
>> --
>> *From*: "Sonny Rajagopalan" 
>> *Sent*: Thursday, January 28, 2016 7:35 PM
>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>> asterisk-users@lists.digium.com>
>> *Subject*: [asterisk-users] Asterisk 13.6.0: Is there a way to create
>> PJSIP users and dialplans programmatically using API
>>
>> Hi,
>>
>> I am using Asterisk 13.6.0 and was wondering if I can programmatically
>> add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk
>> server using API of some sort.
>>
>> Please do let me know.
>>
>>
>>
> With the right Sorcery configuration, you can also use ARI push
> configuration. Creating a PJSIP endpoint, for example, can be done with the
> following:
>
> $ curl -X PUT -H "Content-Type: application/json" -u asterisk:secret -d
> '{"fields": [ { "attribute": "from_user", "value": "alice" }, {
> "attribute": "allow", "value": "!all,g722,ulaw,alaw"}, {"attribute":
> "ice_support", "value": "yes"}, {"attribute": "force_rport", "value":
> "yes"}, {"attribute": "rewrite_contact", "value": "yes"}, {"attribute":
> "rtp_symmetric", "value": "yes"}, {"attribute": "context", "value":
> "default" }, {"attribute": "auth", "value": "alice" }, {"attribute":
> "aors", "value": "alice"} ] }'
> https://localhost:8088/ari/asterisk/config/dynamic/res_pjsip/endpoint/alice
>
> This wiki page describes how this works, as well as how to set it up:
>
> https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration
>
> --
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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