[Freeswitch-users] mod_xml_curl dialplan not requested
Zoltán Szabó
zoell at zoell.us
Sat Nov 29 21:15:31 MSK 2014
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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