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 at 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160923/fe3f83dd/attachment.html>
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
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