Hi,

I have a multi tenant multi domain setup. For some reasons my FS won't try to request the dialplan configuration.

xml_curl:

<configuration name="xml_curl.conf" description="cURL XML Gateway">

  <bindings>

    <binding name="dialplan">

      <param name="gateway-url" value="http://192.168.0.15:8080/ccc-provider/GetDialplan" bindings="dialplan"/>

    </binding>

    <binding name="directory">

      <param name="gateway-url" value="http://192.168.0.15:8080/ccc-provider/GetDirectory" bindings="directory"/>

    </binding>

    <binding name="configuration">

      <param name="gateway-url" value="http://192.168.0.15:8080/ccc-provider/GetConfiguration" bindings="configuration"/>

    </binding>

  </bindings>

</configuration>

My sofia conf that I send through xml_curl upon request:

<document type="freeswitch/xml">

    <section name="configuration">

        <configuration name='sofia.conf' description='sofia endpoint'>

            <global_settings>

                <param name="log-level" value="9"/>

                <param name="debug" value="9"/>

                <param name="auto-restart" value="true"/>

                <param name="debug-presence" value="0"/>

                <param name='accept-blind-reg' value='false'/>

                <param name="sip-trace" value="no"/>

                <param name="sip-capture" value="no"/>

                <param name="auth-calls" value="false"/>

                <param name="dtmf-type" value="rfc2833"/>

            </global_settings>

            <profiles>

                    <profile name='MainCCSIP'>

                        <domains>

                                <domain name="d.1.c.domain.com" alias="false" parse="false"/>

                                <domain name="d.2.c.domain.com" alias="false" parse="false"/>

                        </domains>

                        <settings>

                            <param name="odbc-dsn" value=""/>

                            <param name="manage-presence" value="true"/>

                            <param name="dbname" value="shere_presence"/>

                            <param name="presence-hosts" value="_DISABLED_"/>

                            <param name="send-presence-on-register" value="true"/>

                            <param name="delete-subs-on-register" value="true"/>

                            <param name="caller-id-type" value="rpid"/>

                            <param name="auto-jitterbuffer-msec" value="1000"/>

                            <param name="user-agent-string" value="pbx"/>

                            <param name='dialplan' value='XML'/>

                            <param name="ext-rtp-ip" value="auto"/>

                            <param name="ext-sip-ip" value="auto"/>

                            <param name='rtp-ip' value='auto'/>

                            <param name='sip-ip' value='auto'/>

                            <param name='sip-port' value='5060'/>

                            <param name='nonce-ttl' value=''/>

                            <param name='rtp-timer-name' value='soft'/>

                            <param name='codec-prefs' value='"/>

                            <param name="rtp-timeout-sec" value="300"/>

                            <param name="rtp-hold-timeout-sec" value="1800"/>

                            <param name='rfc2833-pt' value='101'/>

                            <param name='dtmf-duration' value='100'/>

                            <param name="dtmf-type" value=""/>

                            <param name='codec-ms' value='20'/>

                            <param name='accept-blind-reg' value='false'/>

                            <param name='auth-calls' value='false'/>

                            <param name="enable-100rel" value="true"/>

                            <param name="multiple-registrations" value="contact"/>

                            <param name="minimum-session-expires" value="120"/>

                            <param name="mailer-app" value="/usr/local/freeswitch/script/emailvm.py"/>

                            <param name="mailer-app-args" value="-i -d -f voicemail@domain.org -t"/>

                            <param name='accept-blind-reg' value='false'/>

                            <param name='disable-register' value='false'/>

                            <param name='log-auth-failures' value='true'/>

                            <param name='auth-all-packets' value='false'/>

                            <param name="mailer-app" value="/usr/local/freeswitch/scripts/emailvm.py"/>

                            <param name="mailer-app-args" value="-i -d -f voicemail@domain.org -t"/>

                        </settings>

                        <gateways>

                        <gateway name='gateway-1.d.1.c.domain.com'>

                            <variables>

                                <variable name="inbound_var_name"  data="inbound_call=true"  direction="inbound"/>

                                <variable name="outbound_var_name" data="outbound_call=true" direction="outbound"/>

                                <variable name="both_var_name"     data="gw=gwname"/>

                            </variables>

                            <param name='username' value='username'/>

                            <param name='password' value='password'/>

                            <param name="realm" value="sip.voip.com"/>

                            <param name="expire-seconds" value="600"/>

                            <param name='register' value='true'/>

                            <param name="register-transport" value="udp"/>

                            <param name='retry-seconds' value='30'/>

                            <param name="ping" value="60"/>

                            <param name='register' value='true'/>

                        </gateway>

                        </gateways>

                    </profile>

            </profiles>

        </configuration>

    </section>

</document>

My user directory looks like this:

User 1000 at domain 1:

<document type="freeswitch/xml">

    <section name="directory">

        <domain name='d.1.c.domain.com'>

        

        <params>

            <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

        </params>

        

    <variables>

      <variable name="record_stereo" value="true"/>

      <variable name="transfer_fallback_extension" value="operator"/>

      <variable name="user_context" value="d.1.c.domain.com"/>

    </variables>

        

        <groups>

            <group name="d.1.c.domain.com">

                    <users>

                                <user id='1000' mailbox='1000'>

                                    <params>

                                        <param name='password' value='123456'/>

                                        <param name='vm-password' value='123456'/>

                                        <param name="vm-keep-local-after-email" value="true"/>

                                        <param name="vm-message-ext" value="mp3"/>

                                    </params>

                                    <variables>

                                        <variable name="outbound_caller_id_name" value="Unknown"/>

                                        <variable name="outbound_caller_id_number" value="0000000000"/>

                                        <variable name="internal_caller_id_name" value="Unknown"/>

                                        <variable name="user_originated" value="true"/>

                                        <variable name="toll_allow" value="domestic,international,local"/>

                                        <variable name="accountcode" value="1000"/>

                                        <variable name='sip-force-contact' value='nat-connectile-dysfunction'/>

                                        <variable name="record_stereo" value="false"/>

                                        <variable name="sip-allow-multiple-registrations" value="false"/>                                        

                                        <variable name="default_gateway" value="something"/>

                                        <variable name="presence_id" value="1000@d.1.c.domain.com"/>

                                        <variable name="transfer_fallback_extension" value="operator"/>

                                    </variables>

                                </user>

                    </users>

             </group>

          </groups>

        </domain>

    </section>

</document>

User 1000 at domain 2:

<document type="freeswitch/xml">

    <section name="directory">

        <domain name='d.2.c.domain.com'>

        

        <params>

            <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

        </params>

        

    <variables>

      <variable name="record_stereo" value="true"/>

      <variable name="transfer_fallback_extension" value="operator"/>

      <variable name="user_context" value="d.2.c.domain.com"/>

    </variables>

        

        <groups>

            <group name="d.2.c.domain.com">

                    <users>

                                <user id='1000' mailbox='1000'>

                                    <params>

                                        <param name='password' value='654321'/>

                                        <param name='vm-password' value='123456'/>

                                        <param name="vm-keep-local-after-email" value="true"/>

                                        <param name="vm-message-ext" value="mp3"/>

                                    </params>

                                    <variables>

                                        <variable name="outbound_caller_id_name" value="Unknown"/>

                                        <variable name="outbound_caller_id_number" value="0000000000"/>

                                        <variable name="internal_caller_id_name" value="Unknown"/>

                                        <variable name="user_originated" value="true"/>

                                        <variable name="toll_allow" value="domestic,international,local"/>

                                        <variable name="accountcode" value="0"/>

                                        <variable name='sip-force-contact' value='nat-connectile-dysfunction'/>

                                        <variable name="record_stereo" value="false"/>

                                        <variable name="sip-allow-multiple-registrations" value="false"/>                                        

                                        <variable name="default_gateway" value="something"/>

                                        <variable name="presence_id" value="1000@d.2.c.domain.com"/>

                                        <variable name="transfer_fallback_extension" value="operator"/>

                                    </variables>

                                </user>

                    </users>

             </group>

          </groups>

        </domain>

    </section>

</document>

So far the registration for both users in both domains are working and I can register with sip clients. But if I start to dial something, FS won't request for the dialplan. 

Here is what I can see in the cli trace:

freeswitch@dev> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN

tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0)

tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xd47840 from (udp/192.168.0.132:5060) has 972 bytes, veclen = 1

tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xd47840 (972 bytes) from udp/192.168.0.15:5060/sip next=(nil)

nta.c:2880 agent_recv_request() nta: received INVITE sip:123456789@d.2.c.domain.com SIP/2.0 (CSeq 20)

nta.c:3085 agent_recv_request() nta: INVITE (20) going to a default leg

nta.c:1348 set_timeout() nta: timer shortened to 200 ms

nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering

nua_stack.c:899 nh_create() nua: nh_create: entering

nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering

nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering

soa.c:280 soa_clone() soa_clone(static::0xb8f480, 0xb7cea0, 0xb85400) called

soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called

nta.c:4415 nta_leg_tcreate() nta_leg_tcreate(0xdcaf90)

soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0xa98310) called

soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0xa98310, (nil), 0xb54b3b, 433) called

nua_dialog.c:338 nua_dialog_usage_add() nua(0xb85400): adding session usage

tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060

tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060

tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060

tport.c:3492 tport_send_msg() tport_vsend returned 307

nta.c:6789 incoming_reply() nta: sent 100 Trying for INVITE (20)

nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: init -> received, received offer

soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xa98310, [0x7fe32eca6608], [0x7fe32eca6610], [(nil)]) called

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering

2014-11-29 18:31:52.853532 [NOTICE] switch_channel.c:1055 New Channel sofia/MainCCSIP/1000@d.2.c.domain.com [fd50511c-77f5-11e4-b25e-c755bd10628e]

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

2014-11-29 18:31:52.853532 [NOTICE] sofia.c:6946 Hangup sofia/MainCCSIP/1000@d.2.c.domain.com [CS_NEW] [INCOMPATIBLE_DESTINATION]

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:879 nua_respond() nua: nua_respond: entering

nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering

soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called

nua_session.c:2318 nua_invite_server_respond() nua: nua_invite_server_respond: entering

soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0xa98310) called

tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060

tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060

tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060

tport.c:3492 tport_send_msg() tport_vsend returned 892

nta.c:6789 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (20)

nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb85400): removing session usage

nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: received -> terminated

soa.c:356 soa_destroy() soa_destroy(static::0xa98310) called

nta.c:4468 nta_leg_destroy() nta_leg_destroy(0xdcaf90)

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_respond

tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN

tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0)

tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xaa4690 from (udp/192.168.0.132:5060) has 324 bytes, veclen = 1

tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xaa4690 (324 bytes) from udp/192.168.0.15:5060/sip next=(nil)

nta.c:2880 agent_recv_request() nta: received ACK sip:123456789@d.2.c.domain.com SIP/2.0 (CSeq 20)

nta.c:3019 agent_recv_request() nta: ACK (20) is going to INVITE (20)

2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/MainCCSIP/1000@d.2.c.domain.com) Ended

2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close Channel sofia/MainCCSIP/1000@d.2.c.domain.com [CS_DESTROY]

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering

nta.c:4468 nta_leg_destroy() nta_leg_destroy((nil))

nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_destroy

nta.c:1296 agent_timer() nta: timer set next to 1976 ms

The only request is coming from time to time is this:

hostname = [dev]

section = [directory]

tag_name = [domain]

key_name = [name]

key_value = [d.2.c.domain.com]

Event-Name = [REQUEST_PARAMS]

Core-UUID = [d1af88e4-77f4-11e4-b1f2-c755bd10628e]

FreeSWITCH-Hostname = [dev]

FreeSWITCH-Switchname = [dev]

FreeSWITCH-IPv4 = [192.168.0.132]

FreeSWITCH-IPv6 = [::1]

Event-Date-Local = [2014-11-29 18:35:05]

Event-Date-GMT = [Sat, 29 Nov 2014 18:35:05 GMT]

Event-Date-Timestamp = [1417286105312271]

Event-Calling-File = [sofia_reg.c]

Event-Calling-Function = [sofia_reg_parse_auth]

Event-Calling-Line-Number = [2739]

Event-Sequence = [898]

action = [sip_auth]

sip_profile = [MainCCSIP]

sip_user_agent = [qutecom/rev-6653cd8f2afd-trunk]

sip_auth_username = [1000]

sip_auth_realm = [d.2.c.domain.com]

sip_auth_nonce = [6ff7849c-77f6-11e4-b27c-c755bd10628e]

sip_auth_uri = [sip:d.2.c.domain.com]

sip_contact_user = [1000]

sip_contact_host = [192.168.0.15]

sip_to_user = [1000]

sip_to_host = [d.2.c.domain.com]

sip_via_protocol = [udp]

sip_from_user = [1000]

sip_from_host = [d.2.c.domain.com]

sip_call_id = [1172541516@192.168.0.15]

sip_request_host = [d.2.c.domain.com]

sip_auth_response = [6836f04faa9f464b8d78f11e88d7ee5d]

sip_auth_method = [REGISTER]

client_port = [5060]

key = [id]

user = [1000]

domain = [d.2.c.domain.com]

ip = [192.168.0.15]

Any thoughts why is the dialplan not requested?

Thanks,

Zoltan