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


 
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dan at amtelco.com
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PostPosted: Fri Sep 23, 2016 8:32 am    Post subject: [asterisk-users] PJSIP and P-Asserted-Identity Reply with quote

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: 60000
CallerID: calleridname <calleridnumber>
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: 60000
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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jcolp at digium.com
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PostPosted: Fri Sep 23, 2016 8:49 am    Post subject: [asterisk-users] PJSIP and P-Asserted-Identity Reply with quote

Dan Cropp wrote:

<snip>

Quote:

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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dan at amtelco.com
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PostPosted: Fri Sep 23, 2016 8:53 am    Post subject: [asterisk-users] PJSIP and P-Asserted-Identity Reply with quote

Thank you Joshua


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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:

<snip>

Quote:

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

--
_____________________________________________________________________
-- 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
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