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: [email protected] 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: [email protected] [mailto:[email protected]] On Behalf Of Joshua Colp Sent: Monday, October 19, 2015 2:05 PM To: [email protected] 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 functionality It 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 -- _____________________________________________________________________ -- 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
