Re: [asterisk-users] PJSIP Originate
Thanks Joshua -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, March 14, 2018 11:02 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PJSIP Originate On Wed, Mar 14, 2018, at 12:58 PM, Dan Cropp wrote: > I am using AMI Originate to perform a new outbound call. > > The SIP Provider we send the call to wants us to pass the caller id of > the person we are calling for in the Contact header. > > For the AMI Originate, I pass the caller id information data in the > CallerID field. However, this is never being passed through the PJSIP > INVITE header > > Action: Originate > ActionID: S598 > Channel: PJSIP/133@1002 > Exten: createcall > Context: MyContext > Priority: 1 > Timeout: 6 > CallerID: CustomerName <## > > Variable: CALLERID(num- > pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 > Async: true > > Is there a setting that's required on the PJSIP endpoint to allow > overwriting the INVITE packet's Contact header? > Is there something else I am missing to perform this? > > Have a great day! Contact is never used for callerid. The only option available is contact_user on the endpoint to change the Contact username, that's it. -- 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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
Re: [asterisk-users] PJSIP Originate
On Wed, Mar 14, 2018, at 12:58 PM, Dan Cropp wrote: > I am using AMI Originate to perform a new outbound call. > > The SIP Provider we send the call to wants us to pass the caller id of > the person we are calling for in the Contact header. > > For the AMI Originate, I pass the caller id information data in the > CallerID field. However, this is never being passed through the PJSIP > INVITE header > > Action: Originate > ActionID: S598 > Channel: PJSIP/133@1002 > Exten: createcall > Context: MyContext > Priority: 1 > Timeout: 6 > CallerID: CustomerName <## > > Variable: CALLERID(num- > pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 > Async: true > > Is there a setting that's required on the PJSIP endpoint to allow > overwriting the INVITE packet's Contact header? > Is there something else I am missing to perform this? > > Have a great day! Contact is never used for callerid. The only option available is contact_user on the endpoint to change the Contact username, that's it. -- 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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 Originate
I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133@1002 Exten: createcall Context: MyContext Priority: 1 Timeout: 6 CallerID: CustomerName <## > Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 Async: true Is there a setting that's required on the PJSIP endpoint to allow overwriting the INVITE packet's Contact header? Is there something else I am missing to perform this? Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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
