Переменные пользователя не читаются в плане наборе

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Вячеслав Королев

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Dec 7, 2015, 7:32:11 AM12/7/15
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Подскажите пожалуйста по следующему вопросу. Пытаюсь на freeswitch настроить исходящую связь через gateway. При этом необходимо в каждом вызове устанавливать разный АОН в зависимости от переменной пользователя origination_caller_id_name.
Дело в том, что номер А уходящий в gateway не меняется. Почему так происходит?

Настройка пользователя:
  <user id="2003"> <!-- Номер и логин -->
    <params>
        <param name="password" value="12345"/> <!-- пароль -->
        <param name="auth-acl" value="10.10.0.0/16,192.168.1.0/24"/> <!-- с каких сетей можно регистриоваться -->
    </params>
    <variables>
        <variable name="dtmf-type" value="rfc2833"/>  <!-- режим DTMF -->
        <variable name="user_context" value="support_svttk"/> <!-- контекст номера -->
        <variable name="effective_caller_id_name" value="2056598"/> <!-- имя -->
        <variable name="effective_caller_id_number" value="2056598"/> <!-- номер -->
        <variable name="origination_caller_id_name" value="2056598"/>
        <variable name="origination_caller_id_number" value="2056598"/>
        <variable name="sip-force-expires" value="3600"/> <!-- период перерегистрации -->
    </variables>
  </user>

Настройка gateway
   <gateway name="samara-ast1">
     <param name="username" value="xxx"/><!-- -->
     <param name="password" value="yyy"/>
     <param name="realm" value="10.200.104.11"/>
     <param name="from-domain" value="sip.svttk.ru"/>
     <param name="register" value="false"/>
     <param name="caller-id-in-from" value="true"/>
     <param name="sip_cid_type" value="none"/>
     <param name="context" value="support_svttk"/>
   </gateway>

План набора:
    <condition field="destination_number" expression="^([2-9][0-9]{6})$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="export_vars=effective_caller_id_number"/>
        <action application="set" data="effective_caller_id_number=${origination_caller_id_number}"/>
        <action application="set" data="effective_caller_id_name=${origination_caller_id_name}"/>
        <action application="bridge" data="sofia/gateway/samara-ast1/$1"/>
        <action application="hangup"/>
    </condition>

В этом примере, я пытаюсь добиться, что бы в gateway уходил номер FROM  205...@10.200.104.11, а уходит почему-то 20...@10.200.104.11

ros tel

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Dec 7, 2015, 7:41:17 AM12/7/15
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уберите в профилях пользователей

<param name="auth-acl" value="10.10.0.0/16,192.168.1.0/24"/> <!-- с каких сетей можно регистриоваться -->
это всему мешает
 
в диалплане это лишнее
        <action application="set" data="effective_caller_id_number=${origination_caller_id_number}"/>
        <action application="set" data="effective_caller_id_name=${origination_caller_id_name}"/>

т.к. будет применяться из профиля
 <variable name="effective_caller_id_name" value="2056598"/> <!-- имя -->
 <variable name="effective_caller_id_number" value="2056598"/> <!-- номер -->

соответственно в профилях лишние:
 <variable name="origination_caller_id_name" value="2056598"/>
 <variable name="origination_caller_id_number" value="2056598"/>

понедельник, 7 декабря 2015 г., 17:32:11 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 7, 2015, 11:11:57 AM12/7/15
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Попробовал то что посоветовали ros tel.
Ничего не изменилось. В gateway по прежнему уходит номер А 2003, а не 2056598 

Олег Алексеевич

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Dec 7, 2015, 12:11:35 PM12/7/15
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у меня вот так все работает
<include>
  <user id="4211" number-alias="4211">
    <params>
      <param name="password" value="123456"/>
      <param name="vm-password" value="user-choose"/>
      <param name="vm-enabled" value="false"/>
      <param name="vm-email-all-messages" value="true"/>
      <param name="vm-attach-file" value="true"/>
      <param name="vm-keep-local-after-email" value="false"/>
      <param name="vm-mailto" value="a...@mail.ru"/>
      <param name="directory-exten-visible" value="true"/>
      <param name="dial-string" value="{sip_invite_domain=${domain_name},leg_timeout=120,presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>
    <variables>
      <variable name="domain_uuid" value="8y33-bab6-4984-930b-ac14578928de"/>
      <variable name="extension_uuid" value="58y676de-4053-46ff-bde8-4caf6y86r7049"/>
      <variable name="call_group" value="xxx"/>
      <variable name="toll_allow" value="domestic,local,8rt,gorod"/>
      <variable name="accountcode" value="4211"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="4211"/>
      <variable name="effective_caller_id_number" value="4211"/>
      <variable name="outbound_caller_id_name" value="78888869699"/>
      <variable name="outbound_caller_id_number" value="78888869699"/>
      <variable name="emergency_caller_id_number" value="4211"/>
      <variable name="directory_full_name" value="4211"/>
      <variable name="directory-visible" value="true"/>
      <variable name="limit_max" value="5000000"/>
      <variable name="limit_destination" value="09"/>
    </variables>
  </user>
</include>

понедельник, 7 декабря 2015 г., 19:11:57 UTC+3 пользователь Вячеслав Королев написал:

ros tel

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Dec 7, 2015, 10:54:04 PM12/7/15
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значит у вас кроме auth-acl в профиле пользователя, работает acl в sofia-профиле

понедельник, 7 декабря 2015 г., 21:11:57 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 8, 2015, 7:11:15 AM12/8/15
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Проверил возможные параметры от Олега Алексеевича - ничего не помогло.
Вот мой sip профиль с gateway
<profile name="supptechttk">
  <aliases>
  </aliases>
  <gateways>
   <gateway name="samara-ast1">
     <param name="username" value="xxx"/><!-- -->
     <param name="password" value="yyy"/>
     <param name="realm" value="10.200.104.11"/>
     <param name="from-domain" value="sip.svttk.ru"/>
     <param name="register" value="false"/>
     <param name="caller-id-in-from" value="false"/> 
     <param name="contact-params" value="tport=tcp"/>
     <param name="context" value="support_svttk"/>
   </gateway>
  </gateways>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="user-agent-string" value="FreeSWITCH TTK"/> <!-- Название SIP-агента-->
    <param name="caller-id-type" value="rpid"/>
    <!--Диагностика -->
    <param name="debug" value="0"/> <!-- Дебаг отключен -->
    <param name="sip-trace" value="no"/>  <!-- Трассировка отключена -->
    <param name="sip-capture" value="no"/>  <!-- Захват SIP трафика отключен -->
    <param name="log-auth-failures" value="true"/> <!-- Включаем логи неуспешных попыток авторизации-->
    <!-- Параметры DTMF -->
    <param name="rfc2833-pt" value="101"/> <!-- Идентификатор DTMF 2833 в трафике RTP-->
    <param name="dtmf-duration" value="2000"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name=”liberal-dtmf” value=”true”/> <!-- всегда предлагать rfc2833, а принимать и rfc2833, и info dtmf -->
    <!-- Параметры WatchDOG (перезапуск сервера, если он вдруг перестал отвечать) -->
    <param name="watchdog-enabled" value="false"/> <!-- Выключен. -->
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>
    <!-- На каком порту и на каком Ip адресе будет работать профиль-->
    <param name="sip-port" value="5060"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="nonce-ttl" value="60"/> <!-- Параметр TTL-->
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="support_svttk"/>
    <param name="max-proceeding" value="1000"/>
    <param name="hold-music" value="$${hold_music}"/><!-- MOH -->
    <param name="unregister-on-options-fail" value="true"/>
    <param name="all-reg-options-ping" value="true"/>
    <param name="nat-options-ping" value="true"/> 
    <param name="unregister-on-options-fail" value="true"/> 
    <param name="sip-options-respond-503-on-busy" value="true"/>
    <param name="auth-calls" value="false"/>
    <!-- Кодеки и RTP -->
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="disable-transcoding" value="true"/>
    <param name=”inbound-codec-negotiation” value="generous"/> 
    <param name="inbound-late-negotiation" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="auto-jitterbuffer-msec" value="60"/>
    <param name="rtp-timeout-sec" value="300"/> 
    <param name="rtp-hold-timeout-sec" value="1800"/> 
    <!-- Прочие параметры SIP -->
    <param name="enable-timer" value="false"/> 
    <param name="auth-all-packets" value="false"/> 
    <param name="enable-100rel" value="true"/>
    <param name="challenge-realm" value="auto_from"/> 
    <param name="manage-presence" value="false"/> 
    <param name="accept-blind-auth" value="false"/>
    <param name="accept-blind-reg" value="false"/>

    <param name="ws-binding"  value=":5066"/>
    <param name="record-path" value="$${recordings_dir}"/>
    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

  </settings>
</profile>

Как мне кажется ключевой момент в том, что переменной effective_caller_id_number не присвается в процессе набора никакого значения:
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:216 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_ROUTING -> CS_EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/20...@sip.svttk.ru) State ROUTING going to sleep
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/20...@sip.svttk.ru) State EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] mod_sofia.c:196 sofia/supptechttk/20...@sip.svttk.ru SOFIA EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:258 sofia/supptechttk/20...@sip.svttk.ru Standard EXECUTE
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(hangup_after_bridge=true)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [hangup_after_bridge]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(continue_on_fail=true)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [continue_on_fail]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(effective_caller_id_number=)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [effective_caller_id_number]=[UNDEF]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(effective_caller_id_name=)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [effective_caller_id_name]=[UNDEF]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru log(INFO 2003 ----> 9735050)
2015-12-08 15:01:09.171864 [INFO] mod_dptools.c:1692 2003 ----> 9735050

Значение присваивается, только если в диалплане четко прописать например так:
<action application="set" data="effective_caller_id_number=2059568"/>
а вот так не работает
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
Хотя переменная outbound_caller_id_number в описании пользователя в directory определена


ros tel

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Dec 8, 2015, 7:25:08 AM12/8/15
to freeswitch-ru
смотреть лог с самого начала

можно просто удалитить все строки 

<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
из диалплана

не помню есть ли они в ванильном

вторник, 8 декабря 2015 г., 17:11:15 UTC+5 пользователь Вячеслав Королев написал:


Как мне кажется ключевой момент в том, что переменной effective_caller_id_number не присвается в процессе набора никакого значения:
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:216 (sofia/supptechttk/2003@sip.svttk.ru) State Change CS_ROUTING -> CS_EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/2003@sip.svttk.ru) State ROUTING going to sleep
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/2003@sip.svttk.ru) Running State Change CS_EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/2003@sip.svttk.ru) State EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] mod_sofia.c:196 sofia/supptechttk/2003@sip.svttk.ru SOFIA EXECUTE
2015-12-08 15:01:09.171864 [DEBUG] switch_core_state_machine.c:258 sofia/supptechttk/2003@sip.svttk.ru Standard EXECUTE
EXECUTE sofia/supptechttk/2003@sip.svttk.ru set(hangup_after_bridge=true)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/2003@sip.svttk.ru [hangup_after_bridge]=[true]
EXECUTE sofia/supptechttk/2003@sip.svttk.ru set(continue_on_fail=true)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/2003@sip.svttk.ru [continue_on_fail]=[true]
EXECUTE sofia/supptechttk/2003@sip.svttk.ru set(effective_caller_id_number=)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/2003@sip.svttk.ru [effective_caller_id_number]=[UNDEF]
EXECUTE sofia/supptechttk/2003@sip.svttk.ru set(effective_caller_id_name=)
2015-12-08 15:01:09.171864 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/2003@sip.svttk.ru [effective_caller_id_name]=[UNDEF]
EXECUTE sofia/supptechttk/2003@sip.svttk.ru log(INFO 2003 ----> 9735050)
2015-12-08 15:01:09.171864 [INFO] mod_dptools.c:1692 2003 ----> 9735050

Вячеслав Королев

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Dec 8, 2015, 7:54:37 AM12/8/15
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Убрал рекомендуе строки из диал плана.
Лог получился такой
2015-12-08 15:49:10.877978 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/20...@sip.svttk.ru [4eb54ea8-1165-411b-bf5c-afbc3f591ad8]
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_NEW
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:9240 sofia/supptechttk/20...@sip.svttk.ru receiving invite from 10.10.50.240:5060 version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [received][100]
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=Zoiper_user 0 0 IN IP4 10.10.50.240
s=Zoiper_session
c=IN IP4 10.10.50.240
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2015-12-08 15:49:10.877978 [DEBUG] sofia.c:7115 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_NEW -> CS_INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:492 (sofia/supptechttk/20...@sip.svttk.ru) State NEW
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/20...@sip.svttk.ru) State INIT
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:88 sofia/supptechttk/20...@sip.svttk.ru SOFIA INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:40 sofia/supptechttk/20...@sip.svttk.ru Standard INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:48 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_INIT -> CS_ROUTING
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/20...@sip.svttk.ru) State INIT going to sleep
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_ROUTING
2015-12-08 15:49:10.877978 [DEBUG] switch_channel.c:2239 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change DOWN -> RINGING
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/20...@sip.svttk.ru) State ROUTING
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:141 sofia/supptechttk/20...@sip.svttk.ru SOFIA ROUTING
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:166 sofia/supptechttk/20...@sip.svttk.ru Standard ROUTING
2015-12-08 15:49:10.877978 [INFO] mod_dialplan_xml.c:637 Processing 2003 <2003>->9735050 in context support_svttk
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->unloop] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_internal] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [call_internal] destination_number(9735050) =~ /^([1-2][0-9]{3})$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_in] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [call_in] destination_number(9735050) =~ /^(2056598)$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_out] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (PASS) [call_out] destination_number(9735050) =~ /^([2-9][0-9]{6})$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action set(hangup_after_bridge=true)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action set(continue_on_fail=true)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action bridge(sofia/gateway/samara-ast1/9735050)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action hangup()
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:216 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_ROUTING -> CS_EXECUTE
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/20...@sip.svttk.ru) State ROUTING going to sleep
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_EXECUTE
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/20...@sip.svttk.ru) State EXECUTE
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:196 sofia/supptechttk/20...@sip.svttk.ru SOFIA EXECUTE
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:258 sofia/supptechttk/20...@sip.svttk.ru Standard EXECUTE
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(hangup_after_bridge=true)
2015-12-08 15:49:10.877978 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [hangup_after_bridge]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(continue_on_fail=true)
2015-12-08 15:49:10.877978 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [continue_on_fail]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru bridge(sofia/gateway/samara-ast1/9735050)
2015-12-08 15:49:10.877978 [DEBUG] switch_ivr_originate.c:2127 Parsing global variables
2015-12-08 15:49:10.877978 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/9735050 [898b5720-1482-4e7d-9d47-dcc8ae36f182]
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:4765 (sofia/supptechttk/9735050) State Change CS_NEW -> CS_INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/9735050) State INIT
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:88 sofia/supptechttk/9735050 SOFIA INIT
2015-12-08 15:49:10.877978 [DEBUG] sofia_glue.c:1257 sofia/supptechttk/9735050 sending invite version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1449557914 1449557915 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 21036 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:40 sofia/supptechttk/9735050 Standard INIT
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:48 (sofia/supptechttk/9735050) State Change CS_INIT -> CS_ROUTING
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/9735050) State INIT going to sleep
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_ROUTING
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [calling][0]
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/9735050) State ROUTING
2015-12-08 15:49:10.877978 [DEBUG] mod_sofia.c:141 sofia/supptechttk/9735050 SOFIA ROUTING
2015-12-08 15:49:10.877978 [DEBUG] switch_ivr_originate.c:67 (sofia/supptechttk/9735050) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/9735050) State ROUTING going to sleep
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_CONSUME_MEDIA
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:551 (sofia/supptechttk/9735050) State CONSUME_MEDIA
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:551 (sofia/supptechttk/9735050) State CONSUME_MEDIA going to sleep
2015-12-08 15:49:10.997983 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [proceeding][183]
2015-12-08 15:49:10.997983 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=root 33551767 33551767 IN IP4 10.200.104.11
s=Asterisk PBX 1.8.20.1
c=IN IP4 10.200.104.11
t=0 0
m=audio 11982 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 101@8000
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:2898 Set Codec sofia/supptechttk/9735050 PCMA/8000 20 ms 160 samples 64000 bits 1 channels
2015-12-08 15:49:10.997983 [DEBUG] switch_core_codec.c:111 sofia/supptechttk/9735050 Original read codec set to PCMA:8
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4417 Set telephone-event payload to 101@8000
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4473 sofia/supptechttk/9735050 Set 2833 dtmf send payload to 101 recv payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6027 AUDIO RTP [sofia/supptechttk/9735050] 10.200.16.215 port 21036 -> 10.200.104.11 port 11982 codec: 8 ms: 20
2015-12-08 15:49:10.997983 [DEBUG] switch_rtp.c:3788 Starting timer [soft] 160 bytes per 20ms
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6326 sofia/supptechttk/9735050 Set 2833 dtmf send payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6333 sofia/supptechttk/9735050 Set 2833 dtmf receive payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6356 sofia/supptechttk/9735050 Set rtp dtmf delay to 40
2015-12-08 15:49:10.997983 [NOTICE] sofia_media.c:92 Pre-Answer sofia/supptechttk/9735050!
2015-12-08 15:49:10.997983 [DEBUG] switch_channel.c:3460 (sofia/supptechttk/9735050) Callstate Change DOWN -> EARLY
2015-12-08 15:49:10.997983 [INFO] switch_ivr_originate.c:3556 Sending early media
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 101@8000
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:2898 Set Codec sofia/supptechttk/20...@sip.svttk.ru PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2015-12-08 15:49:10.997983 [DEBUG] switch_core_codec.c:111 sofia/supptechttk/20...@sip.svttk.ru Original read codec set to PCMU:0
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4417 Set telephone-event payload to 101@8000
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:4473 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf send payload to 101 recv payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6027 AUDIO RTP [sofia/supptechttk/20...@sip.svttk.ru] 10.200.16.215 port 21374 -> 10.10.50.240 port 8000 codec: 0 ms: 20
2015-12-08 15:49:10.997983 [DEBUG] switch_rtp.c:3788 Starting timer [soft] 160 bytes per 20ms
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6326 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf send payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6333 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf receive payload to 101
2015-12-08 15:49:10.997983 [DEBUG] switch_core_media.c:6356 sofia/supptechttk/20...@sip.svttk.ru Set rtp dtmf delay to 40
2015-12-08 15:49:10.997983 [NOTICE] sofia_media.c:92 Pre-Answer sofia/supptechttk/20...@sip.svttk.ru!
2015-12-08 15:49:10.997983 [DEBUG] switch_channel.c:3460 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change RINGING -> EARLY
2015-12-08 15:49:11.017970 [DEBUG] mod_sofia.c:2320 Ring SDP:
v=0
o=FreeSWITCH 1449557577 1449557578 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 21374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-08 15:49:11.017970 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [early][183]
2015-12-08 15:49:11.017970 [DEBUG] switch_ivr_originate.c:3607 Originate Resulted in Success: [sofia/supptechttk/9735050]
2015-12-08 15:49:11.017970 [DEBUG] switch_core_media.c:9118 sofia/supptechttk/9735050 PAUSE Jitterbuffer
2015-12-08 15:49:11.017970 [DEBUG] switch_core_media.c:9118 sofia/supptechttk/20...@sip.svttk.ru PAUSE Jitterbuffer
2015-12-08 15:49:11.017970 [DEBUG] switch_ivr_bridge.c:1591 (sofia/supptechttk/9735050) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-12-08 15:49:11.017970 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_EXCHANGE_MEDIA
2015-12-08 15:49:11.017970 [DEBUG] switch_core_state_machine.c:542 (sofia/supptechttk/9735050) State EXCHANGE_MEDIA
2015-12-08 15:49:11.017970 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
2015-12-08 15:49:11.037969 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-08 15:49:11.097969 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-08 15:49:11.257969 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [proceeding][180]
2015-12-08 15:49:11.257969 [NOTICE] sofia.c:6852 Ring-Ready sofia/supptechttk/9735050!
2015-12-08 15:49:11.257969 [DEBUG] switch_channel.c:3332 (sofia/supptechttk/9735050) Callstate Change EARLY -> RINGING
2015-12-08 15:49:11.257969 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [completing][200]
2015-12-08 15:49:11.257969 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=root 33551767 33551768 IN IP4 10.200.104.11
s=Asterisk PBX 1.8.20.1
c=IN IP4 10.200.104.11
t=0 0
m=audio 11982 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-12-08 15:49:11.257969 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [ready][200]
2015-12-08 15:49:11.257969 [NOTICE] sofia.c:7655 Channel [sofia/supptechttk/9735050] has been answered
2015-12-08 15:49:11.257969 [DEBUG] switch_channel.c:3759 (sofia/supptechttk/9735050) Callstate Change RINGING -> ACTIVE
2015-12-08 15:49:11.257969 [DEBUG] mod_sofia.c:799 Local SDP sofia/supptechttk/20...@sip.svttk.ru:
v=0
o=FreeSWITCH 1449557577 1449557579 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 21374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-08 15:49:11.257969 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [completed][200]
2015-12-08 15:49:11.257969 [NOTICE] switch_ivr_bridge.c:616 Channel [sofia/supptechttk/20...@sip.svttk.ru] has been answered
2015-12-08 15:49:11.257969 [DEBUG] switch_channel.c:3759 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change EARLY -> ACTIVE
2015-12-08 15:49:11.277969 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [ready][200]
2015-12-08 15:49:11.297980 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-08 15:49:11.317969 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-08 15:49:16.278129 [DEBUG] switch_rtp.c:6859 RTP RECV DTMF 3:960
2015-12-08 15:49:16.278129 [INFO] switch_channel.c:502 RECV DTMF 3:960

Диаллан теперь выглядит так:
  <extension name="call_out">
    <condition field="destination_number" expression="^([2-9][0-9]{6})$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/gateway/samara-ast1/$1"/>
        <action application="hangup"/>
    </condition>
  </extension>
 
В tcpdump видим, что ситуация в принципе не изменилась:
15:52:42.211950 IP (tos 0x0, ttl 64, id 518, offset 0, flags [none], proto UDP (17), length 1076)
    test-centos7.svttk.ru.sip > 10.200.104.11.sip: [bad udp cksum 0x92a3 -> 0x77ea!] SIP, length: 1048
        INVITE sip:973...@10.200.104.11 SIP/2.0
        Via: SIP/2.0/UDP 10.200.16.215;rport;branch=z9hG4bKtvHmSXHS3vpca
        Max-Forwards: 69
        From: "2003" <sip:x...@sip.svttk.ru>;tag=1UmZvm73FUXKF
        To: <sip:973...@10.200.104.11>
        Call-ID: 693f5cae-184d-1234-fd86-000c29a2d7e6
        CSeq: 84453517 INVITE
        Contact: <sip:gw+sama...@10.200.16.215:5060;tport=tcp;transport=udp;gw=samara-ast1>
        User-Agent: FreeSWITCH TTK
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
        Supported: precondition, 100rel, path, replaces
        Allow-Events: talk, hold, conference, refer
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 246
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "2003" <sip:20...@sip.svttk.ru>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1449558654 1449558655 IN IP4 10.200.16.215
        s=FreeSWITCH
        c=IN IP4 10.200.16.215
        t=0 0
        m=audio 20508 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20

ros tel

unread,
Dec 8, 2015, 8:08:53 AM12/8/15
to freeswitch-ru
в sofia-профиле supptechttk стоит

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

вторник, 8 декабря 2015 г., 17:54:37 UTC+5 пользователь Вячеслав Королев написал:
Убрал рекомендуе строки из диал плана.
Лог получился такой
2015-12-08 15:49:10.877978 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/2003@sip.svttk.ru [4eb54ea8-1165-411b-bf5c-afbc3f591ad8]
2015-12-08 15:49:10.877978 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/2003@sip.svttk.ru) Running State Change CS_NEW
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:9240 sofia/supptechttk/2003@sip.svttk.ru receiving invite from 10.10.50.240:5060 version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
2015-12-08 15:49:10.877978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/2003@sip.svttk.ru entering state [received][100]

Вячеслав Королев

unread,
Dec 8, 2015, 10:32:31 AM12/8/15
to freeswitch-ru
Закомментировал переменную в профиле
<param name="auth-calls" value="false"/> 
При этом ничего не изменилось. К сожалению.

ros tel

unread,
Dec 8, 2015, 10:58:47 PM12/8/15
to freeswitch-ru
для применения переменных из профил пользователя FS должен авторизовать его, соответственно на INVITE должен отправить SIP 407, на который клиент должен корректно ответить
у вас авторизация не запрашивается
соответственно либо логи от другого вызова, либо уже что-то наменяли

на авторизацию влияют acl (как в directory так в sofia-профиле) и параметры auth-* 
при попадании в acl авторизация не запрашивается

вторник, 8 декабря 2015 г., 20:32:31 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

unread,
Dec 8, 2015, 11:47:27 PM12/8/15
to freeswitch-ru
Мой профиль:
<profile name="supptechttk">
  <aliases>
  </aliases>
  <gateways>
   <gateway name="samara-ast1">
     <param name="username" value="xxx"/><!-- -->
     <!--<param name="from-user" value="2056598"/>--> <!-- используется для установки поля From при исходящих вызовах через данный gateway -->
     <!--<param name="extension" value="2056598"/>--> <!-- для установки поля To при входящих вызовах с gateway -->
     <param name="password" value="yyy"/>
     <param name="realm" value="10.200.104.11"/>
     <param name="from-domain" value="sip.svttk.ru"/>
     <!--<param name="outbound-proxy" value="10.200.104.11"/>-->
     <!--<param name="expire-seconds" value="60"/>-->
     <param name="register" value="false"/>
     <!--<param name="register-transport" value="udp"/>-->
     <!--<param name="retry-seconds" value="30"/>--> <!-- Количество секунд ожидания перед повтором -->
     <param name="caller-id-in-from" value="false"/> <!-- Использовать АОН входящего вызова в поле from исх вызовов через этот шлюз-->
<!--     <param name="sip_cid_type" value="none"/> -->
     <param name="contact-params" value="tport=tcp"/> <!--extra sip params to send in the contact-->
     <param name="context" value="support_svttk"/>
   </gateway>
  </gateways>

  <domains>
    <domain name="all" alias="false" parse="true"/> <!-- Псевдонимы не создаются, все домены сканируются на предмет подключения gateway -->
  </domains>

  <settings>
    <param name="user-agent-string" value="FreeSWITCH TTK"/> <!-- Название SIP-агента-->
    <param name="caller-id-type" value="rpid"/>
    <!--Диагностика -->
    <param name="debug" value="0"/> <!-- Дебаг отключен -->
    <param name="sip-trace" value="no"/>  <!-- Трассировка отключена -->
    <param name="sip-capture" value="no"/>  <!-- Захват SIP трафика отключен -->
    <param name="log-auth-failures" value="true"/> <!-- Включаем логи неуспешных попыток авторизации-->
    <!-- Параметры DTMF -->
    <param name="rfc2833-pt" value="101"/> <!-- Идентификатор DTMF 2833 в трафике RTP-->
    <param name="dtmf-duration" value="2000"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name=”liberal-dtmf” value=”true”/> <!-- всегда предлагать rfc2833, а принимать и rfc2833, и info dtmf -->
    <!-- Параметры WatchDOG (перезапуск сервера, если он вдруг перестал отвечать) -->
    <param name="watchdog-enabled" value="false"/> <!-- Выключен. -->
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>
    <!-- Главные параметры! На каком порту и на каком Ip адресе будет работать профиль-->
    <param name="sip-port" value="5060"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="nonce-ttl" value="60"/> <!-- Параметр TTL-->
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <!--<param name="apply-nat-acl" value="nat.auto"/>-->
    <!--<param name="local-network-acl" value="localnet.auto"/>-->
    <param name="dialplan" value="XML"/>
    <param name="context" value="support_svttk"/> <!-- Какой dialplan будет включаться при приходе вызова -->
    <param name="max-proceeding" value="1000"/><!-- Максимальное количество обрабатываемых диалогов -->
<!-- <param name="bind-params" value="transport=udp"/> --> <!-- Транспорт только Udp -->
    <param name="hold-music" value="$${hold_music}"/><!-- MOH -->
    <param name="unregister-on-options-fail" value="true"/>
    <param name="all-reg-options-ping" value="true"/>
    <param name="nat-options-ping" value="true"/> <!-- Отправлять периодически параметры OPTION в сторону клиента -->
    <param name="unregister-on-options-fail" value="true"/> <!-- Убивать регистрацию, если пакеты OPTION не имеют ответа -->
    <param name="sip-options-respond-503-on-busy" value="true"/>
    <!--<param name="auth-calls" value="false"/>--><!--Не будет проверки по листам apply-inbound-acl и apply-register-acl -->
<!--    <param name="inbound-reg-force-matching-username" value="true"/> --> <!-- User и authuser должны совпадать-->
    <!-- Кодеки и RTP -->
<!--    <param name="inbound-codec-prefs" value="PCMA, PCMU"/>-->
<!--    <param name="outbound-codec-prefs" value="PCMA, PCMU"/>-->
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="disable-transcoding" value="true"/><!-- Отключить перекодироку. FS будет предлагаеть удаленной стороне только кодеки вызывающей стороры-->
    <param name=”inbound-codec-negotiation” value="generous"/> <!-- Кто главный в выборе кодека generous - удаленная сторона -->
    <param name="inbound-late-negotiation" value="true"/> <!-- Сначала читаем dialplan, потом выбираем кодек -->
    <param name="rtp-timer-name" value="soft"/>
    <param name="auto-jitterbuffer-msec" value="60"/><!-- Буфер для сглаживания rtp jitterа -->
    <param name="rtp-timeout-sec" value="300"/> <!-- Период неактивности RTP, после которого соединение разрывается -->
    <param name="rtp-hold-timeout-sec" value="1800"/>  <!-- Период неактивности RTP, после которого соединение разрывается -->
    <!-- Прочие параметры SIP -->
    <!--<param name="enable-timer" value="false"/>--> <!-- Включить таймеры RFC 4028 SIP Session Timers -->
    <!--<param name="auth-all-packets" value="false"/>--> <!-- Не используется аутентификация при получении всех пакетов (тольо register и invite)-->
    <!--<param name="enable-100rel" value="true"/>-->
    <param name="challenge-realm" value="auto_from"/> <!-- Установка realm, используется поле from в качестве значения SIP realm-->
    <param name="manage-presence" value="false"/> <!--Для отображения состояний (присутсвия) устройств установить «true» -->
    <param name="accept-blind-auth" value="false"/><!-- Запретить слепую аутентификацию -->
    <param name="accept-blind-reg" value="false"/>

    <!-- ДЛЯ БУДУЩЕГО СИПОЛЬЗОВАНИЯ -->
    <!-- for sip over websocket support -->
    <param name="ws-binding"  value=":5066"/>
    <param name="record-path" value="$${recordings_dir}"/>
    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

  </settings>
</profile>

Мой пользователь:
<include>
  <user id="2003"> <!-- Номер и логин -->
    <params>
        <param name="password" value="12345"/> <!-- пароль -->
        <!--<param name="auth-acl" value="10.10.0.0/16,192.168.1.0/24"/>--> <!-- с каких сетей можно регистриоваться -->
        <!--<param name="directory-exten-visible" value="true"/>-->
     </params>
    <variables>
        <variable name="dtmf-type" value="rfc2833"/>  <!-- режим DTMF -->
        <variable name="user_context" value="support_svttk"/> <!-- контекст номера -->
        <variable name="effective_caller_id_name" value="2056598"/> <!-- имя -->
        <variable name="effective_caller_id_number" value="2056598"/> <!-- номер -->
        <variable name="sip-force-expires" value="3600"/> <!-- период перерегистрации -->
        <variable name="outbound_caller_id_number" value="2056598"/>
        <variable name="outbound_caller_id_name" value="2056598"/>
        <!--<variable name="directory-visible" value="true"/>-->
    </variables>
  </user>
</include>

Мой диалплан
<include>
  <extension name="call_out">
    <condition field="destination_number" expression="^([2-9][0-9]{6})$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <!--<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>-->
        <!--<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>-->
        <!--<action application='log' data='INFO ${caller_id_number}  ${destination_number}'/>-->
        <action application="bridge" data="sofia/gateway/samara-ast1/$1"/>
        <action application="hangup"/>
    </condition>
  </extension>
</include>


Лог звонка в fs_cli
2015-12-09 07:39:49.901993 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/20...@sip.svttk.ru [e1f7e63a-8fef-43d8-afab-fce1e062b998]
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_NEW
2015-12-09 07:39:49.901993 [DEBUG] sofia.c:9240 sofia/supptechttk/20...@sip.svttk.ru receiving invite from 10.10.50.240:5060 version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
2015-12-09 07:39:49.901993 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [received][100]
2015-12-09 07:39:49.901993 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=Zoiper_user 0 0 IN IP4 10.10.50.240
s=Zoiper_session
c=IN IP4 10.10.50.240
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2015-12-09 07:39:49.901993 [DEBUG] sofia.c:7115 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_NEW -> CS_INIT
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:492 (sofia/supptechttk/20...@sip.svttk.ru) State NEW
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_INIT
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/20...@sip.svttk.ru) State INIT
2015-12-09 07:39:49.901993 [DEBUG] mod_sofia.c:88 sofia/supptechttk/20...@sip.svttk.ru SOFIA INIT
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:40 sofia/supptechttk/20...@sip.svttk.ru Standard INIT
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:48 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_INIT -> CS_ROUTING
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/20...@sip.svttk.ru) State INIT going to sleep
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_ROUTING
2015-12-09 07:39:49.901993 [DEBUG] switch_channel.c:2239 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change DOWN -> RINGING
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/20...@sip.svttk.ru) State ROUTING
2015-12-09 07:39:49.901993 [DEBUG] mod_sofia.c:141 sofia/supptechttk/20...@sip.svttk.ru SOFIA ROUTING
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:166 sofia/supptechttk/20...@sip.svttk.ru Standard ROUTING
2015-12-09 07:39:49.901993 [INFO] mod_dialplan_xml.c:637 Processing 2003 <2003>->9735050 in context support_svttk
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->unloop] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_internal] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [call_internal] destination_number(9735050) =~ /^([1-2][0-9]{3})$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_in] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (FAIL) [call_in] destination_number(9735050) =~ /^(2056598)$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru parsing [support_svttk->call_out] continue=false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Regex (PASS) [call_out] destination_number(9735050) =~ /^([2-9][0-9]{6})$/ break=on-false
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action set(hangup_after_bridge=true)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action set(continue_on_fail=true)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action bridge(sofia/gateway/samara-ast1/9735050)
Dialplan: sofia/supptechttk/20...@sip.svttk.ru Action hangup()
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:216 (sofia/supptechttk/20...@sip.svttk.ru) State Change CS_ROUTING -> CS_EXECUTE
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/20...@sip.svttk.ru) State ROUTING going to sleep
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/20...@sip.svttk.ru) Running State Change CS_EXECUTE
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/20...@sip.svttk.ru) State EXECUTE
2015-12-09 07:39:49.901993 [DEBUG] mod_sofia.c:196 sofia/supptechttk/20...@sip.svttk.ru SOFIA EXECUTE
2015-12-09 07:39:49.901993 [DEBUG] switch_core_state_machine.c:258 sofia/supptechttk/20...@sip.svttk.ru Standard EXECUTE
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(hangup_after_bridge=true)
2015-12-09 07:39:49.901993 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [hangup_after_bridge]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru set(continue_on_fail=true)
2015-12-09 07:39:49.901993 [DEBUG] mod_dptools.c:1498 SET sofia/supptechttk/20...@sip.svttk.ru [continue_on_fail]=[true]
EXECUTE sofia/supptechttk/20...@sip.svttk.ru bridge(sofia/gateway/samara-ast1/9735050)
2015-12-09 07:39:49.921979 [DEBUG] switch_ivr_originate.c:2127 Parsing global variables
2015-12-09 07:39:49.921979 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/9735050 [ce8220d5-08fb-436e-b1b3-c2d01f4a28b1]
2015-12-09 07:39:49.921979 [DEBUG] mod_sofia.c:4765 (sofia/supptechttk/9735050) State Change CS_NEW -> CS_INIT
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_INIT
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/9735050) State INIT
2015-12-09 07:39:49.921979 [DEBUG] mod_sofia.c:88 sofia/supptechttk/9735050 SOFIA INIT
2015-12-09 07:39:49.921979 [DEBUG] sofia_glue.c:1257 sofia/supptechttk/9735050 sending invite version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1449615411 1449615412 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 20578 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:40 sofia/supptechttk/9735050 Standard INIT
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:48 (sofia/supptechttk/9735050) State Change CS_INIT -> CS_ROUTING
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/9735050) State INIT going to sleep
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_ROUTING
2015-12-09 07:39:49.921979 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [calling][0]
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/9735050) State ROUTING
2015-12-09 07:39:49.921979 [DEBUG] mod_sofia.c:141 sofia/supptechttk/9735050 SOFIA ROUTING
2015-12-09 07:39:49.921979 [DEBUG] switch_ivr_originate.c:67 (sofia/supptechttk/9735050) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/9735050) State ROUTING going to sleep
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_CONSUME_MEDIA
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:551 (sofia/supptechttk/9735050) State CONSUME_MEDIA
2015-12-09 07:39:49.921979 [DEBUG] switch_core_state_machine.c:551 (sofia/supptechttk/9735050) State CONSUME_MEDIA going to sleep
2015-12-09 07:39:50.021989 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [proceeding][183]
2015-12-09 07:39:50.021989 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=root 620890221 620890221 IN IP4 10.200.104.11
s=Asterisk PBX 1.8.20.1
c=IN IP4 10.200.104.11
t=0 0
m=audio 19426 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 101@8000
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:2898 Set Codec sofia/supptechttk/9735050 PCMA/8000 20 ms 160 samples 64000 bits 1 channels
2015-12-09 07:39:50.021989 [DEBUG] switch_core_codec.c:111 sofia/supptechttk/9735050 Original read codec set to PCMA:8
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4417 Set telephone-event payload to 101@8000
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:4473 sofia/supptechttk/9735050 Set 2833 dtmf send payload to 101 recv payload to 101
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:6027 AUDIO RTP [sofia/supptechttk/9735050] 10.200.16.215 port 20578 -> 10.200.104.11 port 19426 codec: 8 ms: 20
2015-12-09 07:39:50.021989 [DEBUG] switch_rtp.c:3788 Starting timer [soft] 160 bytes per 20ms
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:6326 sofia/supptechttk/9735050 Set 2833 dtmf send payload to 101
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:6333 sofia/supptechttk/9735050 Set 2833 dtmf receive payload to 101
2015-12-09 07:39:50.021989 [DEBUG] switch_core_media.c:6356 sofia/supptechttk/9735050 Set rtp dtmf delay to 40
2015-12-09 07:39:50.021989 [NOTICE] sofia_media.c:92 Pre-Answer sofia/supptechttk/9735050!
2015-12-09 07:39:50.021989 [DEBUG] switch_channel.c:3460 (sofia/supptechttk/9735050) Callstate Change DOWN -> EARLY
2015-12-09 07:39:50.041970 [INFO] switch_ivr_originate.c:3556 Sending early media
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 101@8000
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:2898 Set Codec sofia/supptechttk/20...@sip.svttk.ru PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2015-12-09 07:39:50.041970 [DEBUG] switch_core_codec.c:111 sofia/supptechttk/20...@sip.svttk.ru Original read codec set to PCMU:0
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4417 Set telephone-event payload to 101@8000
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:4473 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf send payload to 101 recv payload to 101
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:6027 AUDIO RTP [sofia/supptechttk/20...@sip.svttk.ru] 10.200.16.215 port 18182 -> 10.10.50.240 port 8000 codec: 0 ms: 20
2015-12-09 07:39:50.041970 [DEBUG] switch_rtp.c:3788 Starting timer [soft] 160 bytes per 20ms
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:6326 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf send payload to 101
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:6333 sofia/supptechttk/20...@sip.svttk.ru Set 2833 dtmf receive payload to 101
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:6356 sofia/supptechttk/20...@sip.svttk.ru Set rtp dtmf delay to 40
2015-12-09 07:39:50.041970 [NOTICE] sofia_media.c:92 Pre-Answer sofia/supptechttk/20...@sip.svttk.ru!
2015-12-09 07:39:50.041970 [DEBUG] switch_channel.c:3460 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change RINGING -> EARLY
2015-12-09 07:39:50.041970 [DEBUG] mod_sofia.c:2320 Ring SDP:
v=0
o=FreeSWITCH 1449617808 1449617809 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 18182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-09 07:39:50.041970 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [early][183]
2015-12-09 07:39:50.041970 [DEBUG] switch_ivr_originate.c:3607 Originate Resulted in Success: [sofia/supptechttk/9735050]
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:9118 sofia/supptechttk/9735050 PAUSE Jitterbuffer
2015-12-09 07:39:50.041970 [DEBUG] switch_core_media.c:9118 sofia/supptechttk/20...@sip.svttk.ru PAUSE Jitterbuffer
2015-12-09 07:39:50.041970 [DEBUG] switch_ivr_bridge.c:1591 (sofia/supptechttk/9735050) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-12-09 07:39:50.041970 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/9735050) Running State Change CS_EXCHANGE_MEDIA
2015-12-09 07:39:50.041970 [DEBUG] switch_core_state_machine.c:542 (sofia/supptechttk/9735050) State EXCHANGE_MEDIA
2015-12-09 07:39:50.041970 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
2015-12-09 07:39:50.061981 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-09 07:39:50.141988 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-09 07:39:50.281980 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [proceeding][180]
2015-12-09 07:39:50.281980 [NOTICE] sofia.c:6852 Ring-Ready sofia/supptechttk/9735050!
2015-12-09 07:39:50.281980 [DEBUG] switch_channel.c:3332 (sofia/supptechttk/9735050) Callstate Change EARLY -> RINGING
2015-12-09 07:39:50.301978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [completing][200]
2015-12-09 07:39:50.301978 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=root 620890221 620890222 IN IP4 10.200.104.11
s=Asterisk PBX 1.8.20.1
c=IN IP4 10.200.104.11
t=0 0
m=audio 19426 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-12-09 07:39:50.301978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/9735050 entering state [ready][200]
2015-12-09 07:39:50.301978 [NOTICE] sofia.c:7655 Channel [sofia/supptechttk/9735050] has been answered
2015-12-09 07:39:50.301978 [DEBUG] switch_channel.c:3759 (sofia/supptechttk/9735050) Callstate Change RINGING -> ACTIVE
2015-12-09 07:39:50.301978 [DEBUG] mod_sofia.c:799 Local SDP sofia/supptechttk/20...@sip.svttk.ru:
v=0
o=FreeSWITCH 1449617808 1449617810 IN IP4 10.200.16.215
s=FreeSWITCH
c=IN IP4 10.200.16.215
t=0 0
m=audio 18182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-12-09 07:39:50.301978 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [completed][200]
2015-12-09 07:39:50.301978 [NOTICE] switch_ivr_bridge.c:616 Channel [sofia/supptechttk/20...@sip.svttk.ru] has been answered
2015-12-09 07:39:50.301978 [DEBUG] switch_channel.c:3759 (sofia/supptechttk/20...@sip.svttk.ru) Callstate Change EARLY -> ACTIVE
2015-12-09 07:39:50.321980 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
2015-12-09 07:39:50.321980 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/20...@sip.svttk.ru entering state [ready][200]
2015-12-09 07:39:50.321980 [DEBUG] switch_rtp.c:6640 Correct audio ip/port confirmed.
Type control-D or /exit or /quit or /bye to exit.

freeswitch@internal> /exit

В tcpdump  вижу, что callerid не передается все равно
07:40:42.372284 IP (tos 0x0, ttl 64, id 2594, offset 0, flags [none], proto UDP (17), length 1054)
    test-centos7.svttk.ru.sip > 10.200.104.11.sip: [bad udp cksum 0x928d -> 0xaeef!] SIP, length: 1026
        INVITE sip:973...@10.200.104.11 SIP/2.0
        Via: SIP/2.0/UDP 10.200.16.215;rport;branch=z9hG4bKv4SZvB120U1cr
        Max-Forwards: 69
        From: "2003" <sip:x...@sip.svttk.ru>;tag=Dgm68c2X39v9j
        To: <sip:973...@10.200.104.11>
        Call-ID: d8770b31-18d1-1234-2bbb-000c29a2d7e6
        CSeq: 84481957 INVITE
        Contact: <sip:gw+sama...@10.200.16.215:5060;tport=tcp;transport=udp;gw=samara-ast1>
        User-Agent: FreeSWITCH TTK
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
        Supported: timer, path, replaces
        Allow-Events: talk, hold, conference, refer
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 246
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "2003" <sip:20...@sip.svttk.ru>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1449609800 1449609801 IN IP4 10.200.16.215
        s=FreeSWITCH
        c=IN IP4 10.200.16.215
        t=0 0
        m=audio 26242 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20

По поводу авторизации:
Действительно никаких запросов 407 при приходе INVITE FS в сторону абонента не генерит.
В tcpdump так же заметил, что от FS в сторону клиента идут сообщения SUBSCRIBE, на которые клиент (софтфон Zoyper) отвечает сообщением:
405 Method Not Allowed

Подскажите какие параметры заставят FS запрашивать авторизацию при каждом вызове сообщением 407 и как отключить сообщения SUBSCRIBE, если их клиент не поддерживает.
Я так понимаю сообщением SUBSCRIBE FS пытается подписаться на обновление состояний клиента. Правильно?

ros tel

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Dec 9, 2015, 12:14:23 AM12/9/15
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<!--<param name="auth-calls" value="false"/>--><!--Не будет проверки по листам apply-inbound-acl и apply-register-acl -->
выставить
<param name="auth-calls" value="true"/>
затем
fs_cli -x "sofia profile supptechttk stop" && sleep 30 && fs_cli -x "sofia profile supptechttk start"


примечание 

<!--Не будет проверки по листам apply-inbound-acl и apply-register-acl -->
заблуждение

SUBSCRIBE никаких помех не создают
как отключить отдельная тема

среда, 9 декабря 2015 г., 9:47:27 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 9, 2015, 12:37:45 AM12/9/15
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Выполнил указанные рекомендации.
Теперь действительно при приходе от клиента INVITE происходит запрос 407 на который клиент отвечает.
Далее соединение устанавливается и в tcpdump вижу
    test-centos7.svttk.ru.sip > 10.200.104.11.sip: [bad udp cksum 0x9296 -> 0x1142!] SIP, length: 1035
        INVITE sip:973...@10.200.104.11 SIP/2.0
        Via: SIP/2.0/UDP 10.200.16.215;rport;branch=z9hG4bK65r4ga923XrDB
        Max-Forwards: 69
        From: "2056598" <sip:xxx@sip.svttk.ru>;tag=F1y7S8751UjmF
        To: <sip:973...@10.200.104.11>
        Call-ID: 52dc9314-18d8-1234-47b2-000c29a2d7e6
        CSeq: 84483348 INVITE
        Contact: <sip:gw+sama...@10.200.16.215:5060;tport=tcp;transport=udp;gw=samara-ast1>
        User-Agent: FreeSWITCH TTK
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
        Supported: timer, path, replaces
        Allow-Events: talk, hold, conference, refer
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 246
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "2056598" <sip:205...@sip.svttk.ru>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1449607112 1449607113 IN IP4 10.200.16.215
        s=FreeSWITCH
        c=IN IP4 10.200.16.215
        t=0 0
        m=audio 31712 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
 
То есть поле FROM стало "почти" правильным. Подскажите пожалуйста, как поле FROM привести к виду:
From: "2056598" <sip:2056598@sip.svttk.ru>;tag=F1y7S8751UjmF
ххх - это сейчас значение параметра username в описании настроек gateway

ros tel

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Dec 9, 2015, 12:45:20 AM12/9/15
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в шлюзе samara-ast1
<param name="caller-id-in-from" value="false"/>
заменить на 
<param name="caller-id-in-from" value="true"/>

выполнить
fs_cli -x "sofia profile supptechttk killgw samara-ast1" && sleep 30 && fs_cli -x "sofia profile supptechttk rescan"


среда, 9 декабря 2015 г., 10:37:45 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 9, 2015, 1:13:07 AM12/9/15
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Да, теперь в tcpdump все вроде верно
    test-centos7.svttk.ru.sip > 10.200.104.11.sip: [bad udp cksum 0x929a -> 0xf981!] SIP, length: 1039
        INVITE sip:973...@10.200.104.11 SIP/2.0
        Via: SIP/2.0/UDP 10.200.16.215;rport;branch=z9hG4bK68KgUFtNeNKgK
        Max-Forwards: 69
        From: "2056598" <sip:205...@sip.svttk.ru>;tag=2jNpgNae0KFXB
        To: <sip:973...@10.200.104.11>
        Call-ID: 58744048-18de-1234-47b2-000c29a2d7e6
        CSeq: 84484641 INVITE
        Contact: <sip:gw+sama...@10.200.16.215:5060;tport=tcp;transport=udp;gw=samara-ast1>
        User-Agent: FreeSWITCH TTK
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
        Supported: timer, path, replaces
        Allow-Events: talk, hold, conference, refer
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 246
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "2056598" <sip:205...@sip.svttk.ru>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1449621743 1449621744 IN IP4 10.200.16.215
        s=FreeSWITCH
        c=IN IP4 10.200.16.215
        t=0 0
        m=audio 19668 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
 
Но удаленный gateway при этом стал запрещать соверщать вызов. Ответ от gateway
    10.200.104.11.sip > test-centos7.svttk.ru.sip: [udp sum ok] SIP, length: 460
        SIP/2.0 403 Forbidden
        Via: SIP/2.0/UDP 10.200.16.215;branch=z9hG4bK68KgUFtNeNKgK;received=10.200.16.215;rport=5060
        From: "2056598" <sip:205...@sip.svttk.ru>;tag=2jNpgNae0KFXB
        To: <sip:973...@10.200.104.11>;tag=as35f2ad7d
        Call-ID: 58744048-18de-1234-47b2-000c29a2d7e6
        CSeq: 84484641 INVITE
        Server: Asterisk PBX 1.8.20.1
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0

Попробуй посмотреть на стороне gateway (Asterisk)

ros tel

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Dec 9, 2015, 1:33:29 AM12/9/15
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в пире астреиска

type=peer
host=10.200.104.11
deny=0.0.0.0/0
permit=10.200.104.11/32
insecure=port,invite

среда, 9 декабря 2015 г., 11:13:07 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 9, 2015, 2:44:06 AM12/9/15
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Настройки asterisk
[Freeswitch]
host = 10.200.16.215
context = custumer_freeswitch
type = peer
qualify = yes
disallow = all
allow = alaw
allow = ulaw
dtmfmode = rfc2833
insecure = invite,port
 
Но вызовы не устанавливаются
В логах Asterisk
[Dec  9 11:03:04] NOTICE[15960]: chan_sip.c:23173 handle_request_invite: Failed to authenticate device "2056598" <sip:205...@sip.svttk.ru>;tag=vSgacjg421tvg

Так же пропали работавшие раньше входящие с Asterisk вызовы.
В логах FS
2015-12-09 10:41:34.639106 [WARNING] sofia_reg.c:1760 SIP auth challenge (INVITE) on sofia profile 'supptechttk' for [205...@10.200.16.215] from ip 10.200.104.11

ros tel

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Dec 9, 2015, 3:04:47 AM12/9/15
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потому что в asterisk есть такой пир [2056598]
это уже проблемы asterisk и FS никак не касаются, но можно попрлбовать сделать следующее

в пир [Freeswitch] добавить параметр
trustrpid=yes



в шлюзе samara-ast1
<param name="caller-id-in-from" value="false"/>

тогда пройдет при условии отсутствия пира 2003 на asterisk
и asterisk в CALLERID(number) и CALLERID(name) будет подставлять из заголовка Remote-Party-ID

среда, 9 декабря 2015 г., 12:44:06 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 9, 2015, 7:07:51 AM12/9/15
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Да, все так как Вы описываете.
Исходящая связь через gateway на Asterisk заработала. Все номера передаются правильно.
Но входящие не работают потому что FS спрашивает у Asterisk авторизацию кодом 407.
Не подскажите как ее включить на Asterisk при звонке в сторону FS? 

Вячеслав Королев

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Dec 9, 2015, 7:22:20 AM12/9/15
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Даже, наверное не так,
Подскажите можно ли выключить авторизацию при приходе звонков с определенного IP адреса (от Asterisk) в рамках одного sip_profile?

ros tel

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Dec 9, 2015, 7:36:40 AM12/9/15
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добавить IP asterisk-a в apply-inbound-acl

среда, 9 декабря 2015 г., 17:22:20 UTC+5 пользователь Вячеслав Королев написал:

Вячеслав Королев

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Dec 9, 2015, 11:55:08 AM12/9/15
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Добавил в sip Профиль
<param name="apply-inbound-acl" value="alcgw"/> 
Затем добавил в autoload_configs/acl.conf.xml
    <list name="alcgw" default="deny">
      <node type="allow" cidr="10.200.104.11/32"/>
    </list>

Теперь вызовы входящие через gateway приходят, но до sip_клиента не доходят. Вот лог: 
2015-12-09 19:41:25.735141 [NOTICE] switch_channel.c:1091 New Channel sofia/supptechttk/88469...@10.200.104.11 [6471028f-dfdd-44fd-9af9-279b3e5d1a55]
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_NEW
2015-12-09 19:41:25.735141 [DEBUG] sofia.c:9240 sofia/supptechttk/88469...@10.200.104.11 receiving invite from 10.200.104.11:5060 version: 1.6.5 git 70b8c17 2015-11-20 20:57:50Z 64bit
2015-12-09 19:41:25.735141 [DEBUG] sofia.c:9352 IP 10.200.104.11 Approved by acl "alcgw[]". Access Granted.
2015-12-09 19:41:25.735141 [DEBUG] sofia.c:6750 Channel sofia/supptechttk/88469...@10.200.104.11 entering state [received][100]
2015-12-09 19:41:25.735141 [DEBUG] sofia.c:6760 Remote SDP:
v=0
o=root 1477588519 1477588519 IN IP4 10.200.104.11
s=Asterisk PBX 1.8.20.1
c=IN IP4 10.200.104.11
t=0 0
m=audio 12756 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-12-09 19:41:25.735141 [DEBUG] sofia.c:7115 (sofia/supptechttk/88469...@10.200.104.11) State Change CS_NEW -> CS_INIT
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:492 (sofia/supptechttk/88469...@10.200.104.11) State NEW
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_INIT
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/88469...@10.200.104.11) State INIT
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:88 sofia/supptechttk/88469...@10.200.104.11 SOFIA INIT
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:40 sofia/supptechttk/88469...@10.200.104.11 Standard INIT
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:48 (sofia/supptechttk/88469...@10.200.104.11) State Change CS_INIT -> CS_ROUTING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:516 (sofia/supptechttk/88469...@10.200.104.11) State INIT going to sleep
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_ROUTING
2015-12-09 19:41:25.735141 [DEBUG] switch_channel.c:2239 (sofia/supptechttk/88469...@10.200.104.11) Callstate Change DOWN -> RINGING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/88469...@10.200.104.11) State ROUTING
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:141 sofia/supptechttk/88469...@10.200.104.11 SOFIA ROUTING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:166 sofia/supptechttk/88469...@10.200.104.11 Standard ROUTING
2015-12-09 19:41:25.735141 [INFO] mod_dialplan_xml.c:637 Processing 88469735050 <88469735050>->2057004 in context support_svttk
Dialplan: sofia/supptechttk/88469...@10.200.104.11 parsing [support_svttk->unloop] continue=false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 parsing [support_svttk->call_internal] continue=false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Regex (FAIL) [call_internal] destination_number(2057004) =~ /^([1-2][0-9]{3})$/ break=on-false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 parsing [support_svttk->call_in] continue=false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Regex (PASS) [call_in] destination_number(2057004) =~ /^(2057004)$/ break=on-false
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Action bridge(user/20...@sip.svttk.ru)
Dialplan: sofia/supptechttk/88469...@10.200.104.11 Action hangup()
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:216 (sofia/supptechttk/88469...@10.200.104.11) State Change CS_ROUTING -> CS_EXECUTE
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:532 (sofia/supptechttk/88469...@10.200.104.11) State ROUTING going to sleep
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_EXECUTE
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/88469...@10.200.104.11) State EXECUTE
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:196 sofia/supptechttk/88469...@10.200.104.11 SOFIA EXECUTE
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:258 sofia/supptechttk/88469...@10.200.104.11 Standard EXECUTE
EXECUTE sofia/supptechttk/88469...@10.200.104.11 bridge(user/20...@sip.svttk.ru)
2015-12-09 19:41:25.735141 [DEBUG] switch_ivr_originate.c:2127 Parsing global variables
2015-12-09 19:41:25.735141 [DEBUG] switch_ivr_originate.c:2127 Parsing global variables
2015-12-09 19:41:25.735141 [NOTICE] switch_ivr_originate.c:2762 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2015-12-09 19:41:25.735141 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2015-12-09 19:41:25.735141 [NOTICE] switch_ivr_originate.c:2762 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2015-12-09 19:41:25.735141 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2015-12-09 19:41:25.735141 [INFO] mod_dptools.c:3379 Originate Failed.  Cause: USER_NOT_REGISTERED
2015-12-09 19:41:25.735141 [NOTICE] switch_channel.c:4800 Hangup sofia/supptechttk/88469...@10.200.104.11 [CS_EXECUTE] [USER_NOT_REGISTERED]
2015-12-09 19:41:25.735141 [DEBUG] switch_core_session.c:2796 sofia/supptechttk/88469...@10.200.104.11 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:539 (sofia/supptechttk/88469...@10.200.104.11) State EXECUTE going to sleep
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_HANGUP
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:739 (sofia/supptechttk/88469...@10.200.104.11) Callstate Change RINGING -> HANGUP
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:741 (sofia/supptechttk/88469...@10.200.104.11) State HANGUP
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:431 Channel sofia/supptechttk/88469...@10.200.104.11 hanging up, cause: USER_NOT_REGISTERED
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 480
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:60 sofia/supptechttk/88469...@10.200.104.11 Standard HANGUP, cause: USER_NOT_REGISTERED
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:741 (sofia/supptechttk/88469...@10.200.104.11) State HANGUP going to sleep
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:508 (sofia/supptechttk/88469...@10.200.104.11) State Change CS_HANGUP -> CS_REPORTING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:473 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_REPORTING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:827 (sofia/supptechttk/88469...@10.200.104.11) State REPORTING
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:104 sofia/supptechttk/88469...@10.200.104.11 Standard REPORTING, cause: USER_NOT_REGISTERED
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:827 (sofia/supptechttk/88469...@10.200.104.11) State REPORTING going to sleep
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:499 (sofia/supptechttk/88469...@10.200.104.11) State Change CS_REPORTING -> CS_DESTROY
2015-12-09 19:41:25.735141 [DEBUG] switch_core_session.c:1646 Session 14 (sofia/supptechttk/88469...@10.200.104.11) Locked, Waiting on external entities
2015-12-09 19:41:25.735141 [NOTICE] switch_core_session.c:1664 Session 14 (sofia/supptechttk/88469...@10.200.104.11) Ended
2015-12-09 19:41:25.735141 [NOTICE] switch_core_session.c:1668 Close Channel sofia/supptechttk/88469...@10.200.104.11 [CS_DESTROY]
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:630 (sofia/supptechttk/88469...@10.200.104.11) Running State Change CS_DESTROY
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:640 (sofia/supptechttk/88469...@10.200.104.11) State DESTROY
2015-12-09 19:41:25.735141 [DEBUG] mod_sofia.c:341 sofia/supptechttk/88469...@10.200.104.11 SOFIA DESTROY
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:111 sofia/supptechttk/88469...@10.200.104.11 Standard DESTROY
2015-12-09 19:41:25.735141 [DEBUG] switch_core_state_machine.c:640 (sofia/supptechttk/88469...@10.200.104.11) State DESTROY going to sleep
2015-12-09 19:41:39.055155 [DEBUG] sofia.c:5901 Ping to sip user '20...@sip.svttk.ru' succeeded with code 200 - count 1, state Reachable
Type control-D or /exit or /quit or /bye to exit.

2015-12-09 19:41:58.055150 [DEBUG] sofia.c:5901 Ping to sip user '20...@sip.svttk.ru' succeeded with code 200 - count 2, state Reachable

Как буд-то нет регистрации.
Но согласно вот этим данным регистрация пользователя 2003 есть

freeswitch@internal> sofia status profile supptechttk reg

Registrations:
=================================================================================================
Call-ID:        ZTNmOGVhOTg2OGYwM2Y0MTlhYjgyOGEwNzUyMDkzMGM.
User:           20...@sip.svttk.ru
Contact:        "2003" <sip:20...@10.10.50.240:5060;rinstance=af22998b730004e2;transport=UDP>
Agent:          Zoiper for Windows 2.38 rev.16635
Status:         Registered(UDP)(unknown) EXP(2015-12-09 20:33:27) EXPSECS(3229)
Ping-Status:    Reachable
Ping-Time:      5.11
Host:           test-centos7.svttk.ru
IP:             10.10.50.240
Port:           5060
Auth-User:      2003
Auth-Realm:     sip.svttk.ru
MWI-Account:    20...@sip.svttk.ru

Total items returned: 1
=================================================================================================
 
Подскажите, почему так?
Пробовал так же создавать acl 
<param name="apply-register-acl" value="acluser"/>
Но ничего не изменилось.

Вячеслав Королев

unread,
Dec 9, 2015, 11:07:29 PM12/9/15
to freeswitch-ru
С последним вопросом сам разобрался. У меня пользователь регистрировался не в том домене.
Спасибо большое за помощь. 
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