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 [[email protected]] 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="${voicemail_accou...@${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/[email protected] 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/[email protected]] 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/[email protected]! 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/[email protected]! 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/[email protected] [BREAK] 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/[email protected] entering state [early] 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [[email protected]] 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 l...@8000hz 1 channels 20ms 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/[email protected] 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/[email protected] [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/[email protected] [KILL] 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/[email protected] [BREAK] 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/[email protected]) State EXECUTE going to sleep 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/[email protected]) Running State Change CS_HANGUP 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/[email protected]) State HANGUP 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/[email protected] 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/[email protected] Standard HANGUP, cause: NORMAL_CLEARING 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/[email protected]) 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/[email protected]) 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/[email protected]) Ended 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/[email protected] [CS_HANGUP] -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
