[Freeswitch-users] mod_xml_curl dialplan not requested
Zoltán Szabó
zoell at zoell.us
Sat Nov 29 22:23:48 MSK 2014
One addition to this is my modules conf:
<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_logfile"/>
<load module="mod_xml_curl"/>
<load module="mod_dialplan_xml"/>
<load module="mod_enum"/>
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<load module="mod_commands"/>
<load module="mod_db"/>
<load module="mod_dptools"/>
<load module="mod_expr"/>
<load module="mod_esf"/>
<load module="mod_fsv"/>
<load module="mod_valet_parking"/>
<load module="mod_httapi"/>
<load module="mod_g723_1"/>
<load module="mod_g729"/>
<load module="mod_amr"/>
<load module="mod_h26x"/>
<load module="mod_vp8"/>
<load module="mod_b64"/>
<load module="mod_sndfile"/>
<load module="mod_native_file"/>
<load module="mod_local_stream"/>
<load module="mod_tone_stream"/>
<load module="mod_say_en"/>
</modules>
</configuration>
2014-11-29 18:15 GMT+00:00 Zoltán Szabó <zoell at zoell.us>:
> 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at d.2.c.domain.com) Ended
> 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close
> Channel sofia/MainCCSIP/1000 at 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 at 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
>
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