Hi Joshua If i put the default_user option per endpoint would it work?? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality? Thanks<div> </div><div> </div><!-- originalMessage --><div>-------- Original message --------</div><div>From: Joshua Colp <jcolp at digium.com> </div><div>Date: 2015/10/19 13:03 (GMT+02:00) </div><div>To: asterisk-users at lists.digium.com </div><div>Subject: Re: [asterisk-users] Modify Contact in PJsip </div><div> </div>On 15-10-19 07:41 AM, Andrew Colin wrote:> Hi Guys > > We are using the wizard to configure our pjsip trunk(see below) > > How do we get this setting to work > > contact_user=username > > We want to change the contact field in the sip invite to display the > username of the trunk >The Contact header can not currently be modified on a per-endpoint basis and takes its values from the generated From header. On a global scale it could be controlled using the default_user global option. Otherwise there's no real way without adding explicit support for 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151019/c198dd3e/attachment.html>
On 15-10-19 08:17 AM, Andrew Colin wrote:> Hi Joshua > > If i put the default_user option per endpoint would it work?No, it's a global only option.> > So what exactly does the contact_user option do?It sets the Contact user in an outbound registration so that the URI dialed by the remote SIP server may contain that user (or may not, depending on their configuration/deployment).> > I know that in freeswitch there is the option extension-in-contact. > We basically need to achieve the same functionalityIt would require modifying the code and adding support. -- 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
Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10?591 4600 Email:? andrew at convergedgroup.net Web: ?http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you. E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, October 19, 2015 2:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Modify Contact in PJsip On 15-10-19 08:17 AM, Andrew Colin wrote:> Hi Joshua > > If i put the default_user option per endpoint would it work?No, it's a global only option.> > So what exactly does the contact_user option do?It sets the Contact user in an outbound registration so that the URI dialed by the remote SIP server may contain that user (or may not, depending on their configuration/deployment).> > I know that in freeswitch there is the option extension-in-contact. > We basically need to achieve the same functionalityIt would require modifying the code and adding support. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users