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

[Dan Cropp]

Thank you Joshua


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

>
> 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