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

Brian West brian at freeswitch.org
Tue Dec 30 10:47:24 PST 2008


I would update to the new method using groups

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="arbitrary stuff here">
	<domain name="foo.com">
		<groups>
       		<group name="default">
         		<users>
          			<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>

         		</users>
       		</group>
		</groups>
	</domain>
</section>
</document>


/b


On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:

> you need to add something similar to the following to your directory
> request:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">
>
>
> -Ray
>
>
>
>
> can_man at gmx.de wrote:
>> 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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