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    <div class="moz-cite-prefix">07.03.2015 1:21, Kevin Harwell пишет:<br>
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    <blockquote
cite="mid:CAM-yhnk6NYZnGiHcJwP7ysbA3MR5X99Sc24pjaW80MxDMStz1g@mail.gmail.com"
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          <div class="gmail_quote">On Fri, Mar 6, 2015 at 3:46 PM,
            Dmitriy Serov <span dir="ltr"><<a moz-do-not-send="true"
                href="mailto:serov.d.p@gmail.com" target="_blank">serov.d.p@gmail.com</a>></span>
            wrote:<br>
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                <div>07.03.2015 0:24, Kevin Harwell пишет:<br>
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                    <blockquote type="cite">
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                        <div class="gmail_extra">
                          <div class="gmail_quote">On Fri, Mar 6, 2015
                            at 2:06 PM, Dmitriy Serov <span dir="ltr"><<a
                                moz-do-not-send="true"
                                href="mailto:serov.d.p@gmail.com"
                                target="_blank">serov.d.p@gmail.com</a>></span>
                            wrote:<br>
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                              style="margin:0px 0px 0px
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                                Hello.<br>
                                <br>
                                Asterisk 13.2.<br>
                                I transfer configs from chan_sip to
                                res_pjsip.<br>
                                In chan_sip i have
                                "match_auth_username=yes" and have
                                nothing in pjsip.<br>
                                <br>
                                I have a lot of endpoints and
                                registrations on same SIP server. And
                                it's problem in pjsip now. Is not it?<br>
                                <br>
                                I requesting to add new value for
                                endpoint option identify_by. The value
                                'uri'.<br>
                                Simple config (cutted):<br>
                                <br>
                                <pre style="margin:0px 0px 1.4em;padding:0.7em 1em;font-family:Consolas,"Andale Mono WT","Andale Mono","Bitstream Vera Sans Mono","Nimbus Mono L",Monaco,"Courier New",monospace;font-size:14px;direction:ltr;text-align:left;color:rgb(51,51,51);border-radius:2px;overflow:auto;word-wrap:normal;border:1px solid rgb(204,204,204);font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:19.6px;text-indent:0px;text-transform:none;word-spacing:0px;background-color:rgb(251,250,249)">[siptrunk]
type=registration
transport=udp-transport
outbound_auth=siptrunk
server_uri=sip:<a moz-do-not-send="true" href="http://sip.example.com" target="_blank">sip.example.com</a>
<a moz-do-not-send="true" href="mailto:client_uri=sip:1234567890@sip.example.com" target="_blank">client_uri=sip:1234567890@sip.example.com</a>
retry_interval=60
contact_user=siptrunk-in

[siptrunk-in]
type=endpoint
transport=udp-transport
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=siptrunk
aors=siptrunk
identify_by=uri
</pre>
                                <br>
                                Registration section has option
                                "contact_user". Incoming call from this
                                registration will be INVITE
                                sip:siptrunk-in@....<br>
                                I offer to change
                                res_pjsip_endpoint_identifier_user to
                                realize endpoint identification by sip
                                uri.<br>
                                <br>
                                I think it will be usefull.<br>
                                <br>
                                P.S. i hope issues will be rejected: <a
                                  moz-do-not-send="true"
                                  href="https://issues.asterisk.org/jira/browse/ASTERISK-22306"
                                  target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-22306</a>
                                and SWP-6069  <span
style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:20px;text-align:start;text-indent:0px;text-transform:none;white-space:nowrap;word-spacing:0px;display:inline!important;float:none;background-color:rgb(230,230,230)"><span></span></span><span
                                  title="SWP-6069 res_pjsip endpoint
                                  config object's 'identify_by' option
                                  needs cleanup, removal or other
                                  modification"
style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:20px;text-align:start;text-indent:0px;text-transform:none;white-space:nowrap;word-spacing:0px;background-color:rgb(230,230,230)"><br>
                                  <span><font color="#888888"> <br>
                                      <br>
                                    </font></span></span><span><font
                                    color="#888888">Dmitriy Serov </font></span></div>
                              <br>
                              -<br clear="all">
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                        <div class="gmail_extra">I believe what you are
                          looking for is already available. See the
                          "identify" type (type=identify) section that
                          is in the pjsip.conf file and the "identify"
                          option for endpoints. These allow you to
                          identify and endpoint by IP address. <br>
                          <br>
                          For more information see the pjsip.conf.sample
                          file.  Also take a look at configuring
                          Asterisk for res_pjsip [1] specifically the
                          part about configuring endpoint identification
                          by IP address [2]. If you run into problems
                          more information can also be found in the
                          res_pjsip troubleshooting guide [3],
                          specifically the section on "identify by IP
                          address"<br>
                          <br>
                          [1] <a moz-do-not-send="true"
                            href="https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip"
                            target="_blank">https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip</a><br>
                          [2] <a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip"
                            target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip</a><br>
                          [3] <a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide"
                            target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide</a><br>
                          <br>
                        </div>
                        <div class="gmail_extra">Hope that helps,<br>
                        </div>
                        <div class="gmail_extra"><br>
                        </div>
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                Thank you for answer. But...<br>
                ones again: I have a lot of endpoints and registrations
                on same SIP server. And it's problem in pjsip now. Is
                not it?<br>
                <br>
                Simple Example. I have two trunks with their own
                credentials (and did) to the same sip server:<br>
                - for home<br>
                - for bussiness<br>
                <br>
                [home-example.com-endpoint]<br>
                <br>
                [bussiness-example.com-endpoint]<br>
                <br>
                [home-example.com-registration]<br>
                contact_user=home-example.com-endpoint<br>
                <br>
                [bussiness-example.com-registration]<br>
                contact_user=bussiness-example.com-endpoint<br>
                <br>
                ;and ok... i wrote identify by IP section<br>
                [example.com-identify]<br>
                type=identify<br>
                match=<a moz-do-not-send="true"
                  href="http://example.com" target="_blank">example.com</a><br>
                endpoint= ???<br>
                <br>
                It is very! important for me to know what trunk passes
                through the incoming call: home or bussiness.<br>
                1. Identify by IP. Do you have answer?<br>
                2. Identify by username. What? I can't make endpoints to
                all of my contacts.<br>
                <br>
                Ok. I can use contact_user in registraction and route
                incoming call by INVITE uri.<br>
                Can i?<span class="HOEnZb"><font color="#888888"><br>
                    <br>
                    Dmitriy Serov<br>
                    <br>
                  </font></span></div>
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          <br>
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        <div class="gmail_extra">I don't think I fully understand the
          scenario, but if you have different named endpoints
          originating from the same address (or even different
          addresses) then these can be identified by the username
          portion of the sip uri. However, if you have endpoints for
          instance with the same name, but different addresses then
          these can be distinguished by using the identity type.<br>
          <br>
        </div>
        <div class="gmail_extra">Your scenario looks like the first
          option. The endpoint names are different, but the address is
          the same, so identification based on the username should be
          sufficient.<br>
          <br>
        </div>
        <div class="gmail_extra">However, if you have a mix of both
          types on your system, for instance multiple endpoints on two
          different systems (IP addresses) with the same names, then I
          am unsure how you would select the correct endpoint even while
          attempting to "identify_by" uri.<br clear="all">
        </div>
        <div class="gmail_extra"><br>
          -- <br>
          <div class="gmail_signature">
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              <pre style="padding:2px;border:1px solid rgb(114,99,77);background-color:rgb(238,238,238);color:rgb(0,0,0);overflow:auto">Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: <a moz-do-not-send="true" href="http://digium.com" target="_blank">http://digium.com</a> & <a moz-do-not-send="true" href="http://asterisk.org" target="_blank">http://asterisk.org</a></pre>
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    </blockquote>
    <br>
    I have different named endpoints originating from the same address.<br>
    >>then these can be identified by the username portion of the
    sip uri<br>
    ok. A little more examples.<br>
    <br>
    case 1. Device!<br>
    we have: endpoint in config and registered device.<br>
    incoming call. In this case identification makes by username portion
    of sip uri "from". And we know endpoint.<br>
    Excellent!<br>
    <br>
    case 2. Trunk!<br>
    we have: endpoint and registration of trunk1 to sip server
    <a class="moz-txt-link-rfc2396E" href="mailto:user@example.com">"user@example.com"</a> in config.<br>
    Registration section have option contact_user. I can fill it by
    usefull value. For example: "201"<br>
    and now... <a class="moz-txt-link-abbreviated" href="mailto:somebody@example.com">somebody@example.com</a> makes call to <a class="moz-txt-link-abbreviated" href="mailto:user@example.com">user@example.com</a><br>
    i have incoming call to my server:<br>
    INVITE sip:201@<br>
    to: 201<br>
    from: <a class="moz-txt-link-abbreviated" href="mailto:somebody@example.com">somebody@example.com</a><br>
    Question: how can i identify my endpoint?<br>
    <br>
    P.S. discussion of this issue goes in IRC #asterisk-dev now. And
    also difficult :(<br>
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