On Tue, Sep 26, 2017, at 05:53 PM, marek cervenka wrote: > Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): > > On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > >> hi, > >> > >> i want use asterisk+pjsip as voip client with multiple registrations > >> (perf testing) > >> > >> i'm using this example configuration for one account > >> > >> [308] > >> type=registration > >> outbound_auth=308 > >> server_uri=sip:3...@example.com:5060 > >> client_uri=sip:3...@example.com:5060 > >> > >> [308](auth-userpass) > >> username=308 > >> password=pass > >> > >> [308](aor-single-reg) > >> contact=sip:example.com:5060 > >> > >> [308](endpoint-basic) > >> outbound_auth=308 > >> aors=308 > >> > >> [308] > >> type=identify > >> endpoint=308 > >> match=example.com > >> > >> > >> my problem is contact on the other side (is same for all endpoints) > >> > >> Addr->IP : 1.1.1.1:5060 > >> Reg. Contact : sip:s@1.1.1.1:5060 > >> > >> all incoming calls from PBX to my Asterisk are routed to only one > >> account (because of same ip address/port ?) > >> > >> how can i specify different source port or different contact address for > >> asterisk pjsip client with registration? > > The "contact_user" option configures the user portion of the Contact > > that is sent in the REGISTER. You can set it to a different value for > > each registration. > > ok i have this configuration now > client - asterisk+pjsip (public ip 1.1.1.1) > pjsip/307 > pjsip/308 > > server - asterisk+chan_sip (public ip 2.2.2.2) > sip/307 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:307@1.1.1.1:5060 > > sip/308 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:308@1.1.1.1:5060 > > > now, every call from server to client is received through pjsip/307 . > but i need receive call for pjsip/308 through registration of pjsip/308. > is it possible? > is it possible configure different source port other than 5060?
There is no ability to match to an endpoint currently based on the transport traffic comes in on. You can try enabling the line option[1] which may allow the inbound calls to be directed to an alternative endpoint. If this doesn't work you'll need to match all incoming to a single endpoint and direct things appropriately in the dialplan based on the dialed number. [1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- 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
