[asterisk-users] Coming to AstriCon? Check out these presentations by Asterisk Team Members

2016-09-23 Thread Asterisk Development Team 
Check out these 4 presentations by members of the Asterisk Development Team:


*Deploying Digium Phones the easy way with DPMA*
How to configure and manage your phones over a variety of networks,
protocols and distributions.
Tuesday 5:00PM
Presented by: Scott Griepentrog

*PJSIP: Tuning for Performance*
Learn how tune the Asterisk PJSIP channel driver for a high volume
environment.  Includes discussions about, and examples of, configuring
realtime database access, the use of caches and other configure options and
distribution of workload.
Wednesday 3:25PM
Presented by: Mark Michelson and George Joseph

*Asterisk Testing for Your Deployments and the Project*
How the Asterisk testsuite helps to uncover issues before they appear in
your deployment and how you can use its fundamentals to test the unique
parts of your deployment.
Thursday 11:45AM
Presented by: Joshua Colp and Scott Griepentrog

*AstriCon Wrap Up*
Asterisk: Past, Present, and Future
Thursday 3:10PM
Presented by: Matt Fredrickson and Matt Jordan
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk Radius CDR

2016-09-23 Thread Ahmed Munir 
Hi,

I've recently setup Asterisk with Radius CDR by following the document:
https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.

The issue currently I'm facing is after turning on the debug getting
message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR
not recorded!

I've checked and grant access 666 to radiusclient config files: servers &
dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that
/var/run/radius.seq is not getting updated.


Further added, in asterisk CLI while running command: cdr show status
getting results below;

Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   No
  Log congestion: No

* Registered Backends
  ---
cdr-syslog
Adaptive ODBC
cdr-custom
csv
radius


Please advise if I may missed any steps.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

2016-09-23 Thread Matthew Jordan 
On Wed, Sep 21, 2016 at 9:27 AM, Amit Patkar  wrote:
> Thanks Mathew. I understand that there is no coordination between AsyncAGI &
> AMI.
> Is there any dial plan function which can tell us if there is active AMI
> session?
>

Assuming you know the client name (login name), you can use the
AMI_CLIENT [1] dialplan function to retrieve the number of sessions
they have currently established.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_AMI_CLIENT

-- 
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] PJSIP and P-Asserted-Identity

2016-09-23 Thread Dan Cropp 
Thank you Joshua


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Friday, September 23, 2016 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP and P-Asserted-Identity

Dan Cropp wrote:



>
> If no caller id is present, calls go through IPitimi to my cell phone.
> However, if caller id is present, the P-Asserted-Identity is the 
> caller id. Based on conversations with IPitimi and some other SIP 
> products, this is incorrect. The P-Asserted-Identity should be the 
> from_user at from_domain and the From and Contact should be the Caller 
> Id provided information.

This is the opposite of how most ITSPs and deployments expect things to 
operate, which is why you're seeing the behavior you are. It's not written to 
behave like this. Your best bet is to do as you've mentioned and manually 
manipulate things. This is also, I think, the first time I've ever heard of a 
company wanting it to behave precisely like that.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
www.digium.com & www.asterisk.org


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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] PJSIP and P-Asserted-Identity

2016-09-23 Thread Joshua Colp 

Dan Cropp wrote:





If no caller id is present, calls go through IPitimi to my cell phone.
However, if caller id is present, the P-Asserted-Identity is the caller
id. Based on conversations with IPitimi and some other SIP products,
this is incorrect. The P-Asserted-Identity should be the from_user at
from_domain and the From and Contact should be the Caller Id provided
information.


This is the opposite of how most ITSPs and deployments expect things to 
operate, which is why you're seeing the behavior you are. It's not 
written to behave like this. Your best bet is to do as you've mentioned 
and manually manipulate things. This is also, I think, the first time 
I've ever heard of a company wanting it to behave precisely like that.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP and P-Asserted-Identity

2016-09-23 Thread Dan Cropp 
I am working with a customer and their SIP provider is IPitimi.

The customer needs to sometimes provide various caller id number for the calls 
going to IPitimi.  They are processing calls for multiple businesses who want 
their caller id to show up.

When no caller id is provided, the From must be the DID at ipitimi ip address 
and caller id is DID at customer IP address.

When caller id is present, the From must be the caller id number at ipitimi ip 
address and caller id is DID at customer IP address.  The P-Asserted-Identity 
must be the DID at ipitimi ip address.

For the endpoint, I have...
from_domain = ipitimi ip address
from_user = DID
send_pai = yes

If no caller id is present, calls go through IPitimi to my cell phone.  
However, if caller id is present, the P-Asserted-Identity is the caller id.  
Based on conversations with IPitimi and some other SIP products, this is 
incorrect.  The P-Asserted-Identity should be the from_user at from_domain and 
the From and Contact should be the Caller Id provided information.


I am Originating the calls using AMI
Sample with caller id...
Action: Originate
ActionID: 1234
Channel: PJSIP/numbertocall@IPitimi
Exten: myexten
Context: Test
Priority: 1
Timeout: 6
CallerID: calleridname 
Variable: CALLERID(num-pres)=allowed_passed_screen
Async: true


Sample without caller id...
Action: Originate
ActionID: 1234
Channel: PJSIP/numbertocall@IPitimi
Exten: myexten
Context: Test
Priority: 1
Timeout: 6
Async: true


Am I missing a setting for the endpoint which places the from_user at 
from_domain in the PAI when caller id is present in the Originate?
Or do I need to remove the from_user setting and have the code do the work of 
determining the from user and setting the PJSIP_Header for PAI when necessary?


Thank you
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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