VoIP Mailing List Archives

[can_man at gmx.de]

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@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@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@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@sipgate.de!
2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous@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@sipgate.de [BREAK]
2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous@sipgate.de entering state [early]


2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315@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@8000hz 1 channels 20ms
2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous@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@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@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@sipgate.de [BREAK]
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous@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@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@sipgate.de) State HANGUP
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous@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@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@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@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@sipgate.de) Ended
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous@sipgate.de [CS_HANGUP]
--
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[brian at freeswitch.org]

what svn rev are you on?

/b

On Dec 30, 2008, at 12:10 PM, can_man@gmx.de wrote:

[intralanman at freeswi...]

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@gmx.de wrote:

[brian at freeswitch.org]

and end that with

</section>
</document>



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

[brian at freeswitch.org]

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:

[can_man at gmx.de]

Hello,

thank you for your answers. I have added the start and end tags to my xml, but nothing has changed.
However, the "groups XML" did work with my server's IP as: <domain name="192.168.178.22"> - thank you Brian.

If someone can shade some light into this quote from 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

I will re-write the whole "directory" section on the wiki. For now I will add the "group" reply.

Thank you,
Phil

Ps: if it is still of interest, my svn version is:

URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org/svn
Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2
Revision: 10988
Node Kind: directory
Schedule: normal
Last Changed Author: brian
Last Changed Rev: 10983
Last Changed Date: 2008-12-29 06:27:53 +0100 (Mon, 29 Dec 2008)

[brian at freeswitch.org]

Thank you... it needed to be updated

btw we now have a group/ endpoint.. so you can call group/sales@domain
and ring everyone in the group. Go check out the default config it
has use examples.

/b

On Dec 30, 2008, at 1:08 PM, can_man@gmx.de wrote: