[Freeswitch-users] voicemail - Can't find user

Tue Dec 30 10:10:27 PST 2008

Hello,

I am trying to get voicemail to run through xml curl, but I get the following error:

2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]

In order to setup user 315 I reply the following to the "directory" request of xml curl:

<user id="315" mailbox="315">
<params>
<param name="password" value="1234"/>
<param name="vm-password" value="0000"/>
</params>
<variables>
<variable name="accountcode" value="315"/>
<variable name="user_context" value="default"/>
<variable name="vm_extension" value="315"/>
<variable name="max_calls" value="1"/>
<variable name="fail_over" value="415"/>
<variable name="cringback" value="us-ring"/>
</variables>
</user>

And in order to send the call to voicemail I do:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="public">
<extension name="test10000">
<condition field="destination_number" expression="^(10000)$">
<action application="voicemail" data="default $${domain} 315"/>
</condition>
</extension>
</context>
</section>
</document>

Do I maybe have to add the user also at another location? 
Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
And I do the same, I respond always with the directory response above. Is there a better practice?

It would be great if someone could point out my error. 

Thank you,
Phil

my voicemail conf looks like this:

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="wav"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="3"/>
      <param name="max-record-len" value="300"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <!--<param name="storage-dir" value="/tmp"/>-->
      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
      <!--<param name="record-comment" value="Your Comment"/>-->
      <!--<param name="record-title" value="Your Title"/>-->
      <!--<param name="record-copyright" value="Your Copyright"/>-->
    </profile>
  </profiles>
</configuration>

the debug output:

2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous at sipgate.de
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20
2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP:
v=0
o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
s=FreeSWITCH
c=IN IP4 89.49.116.108
t=0 0
m=audio 61125 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [early]

2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]

2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en]
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en)
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file
2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NORMAL_CLEARING
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Ended
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]
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