[Freeswitch-users] mod_xml_curl dialplan not requested
Steven Ayre
steveayre at gmail.com
Sun Nov 30 15:43:05 MSK 2014
Your call is being rejected with 488 Not Acceptable Here. That means a
codec error - the client isn't offerring any of the codecs you've
configured FreeSWITCH to accept. Such calls get rejected immediately,
before hitting the dialplan (unless using late-negotiation).
Incidentally, don't use the sofia stack logging (sofia loglevel). Use the
FreeSWITCH logging ("/log debug" in fs_cli). The sofia stack is rather
verbose and unreadable, and doesn't tell you anything about what FreeSWITCH
is doing at the higher level. Looking inside the stack is something to do
if the FreeSWITCH log isn't showing you the error. The FreeSWITCH log'll
show the SDP and the codec negotiation, so you'll see the error there.
On 29 November 2014 at 18:15, Zoltán Szabó <zoell at zoell.us> wrote:
> 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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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
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