First of all I apologise for my long post. In the meantime I reworked the simple js as suggested (adding the domain or using transfer instead of bridge) but yet the script doesn't work. Let me add some other info. I just cloned the '5000' demo_ivr and reworked a little to reach my target. Next I added: <!-- a sample IVR --> <extension name="ivr_demo_alarm"> <condition field="destination_number" expression="^118$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr_alarm"/> </condition> </extension> into the default.xml dialplan. Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) and than I dial '118'. The IVR works fine. Next from the console: originate sofia/my.freeswitch.address/1001 118 I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) That is the log: ========================================================= originate sofia/10.0.0.33/1001 118 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] API CALL [originate(sofia/10.0.0.33/1001 118)] output: +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 freeswitch@Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->118 in context default 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 33 (sofia/internal/1001) Ended 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ==================================================== Using the js instead it looks like if the bridge commands looks for a REGISTERED user at ext. 118, and so it fails. Using the transfer option, as suggested, the Ekiga user (1001) get to be connected to the ext 118, bur the connection drops immediately after. ==================================================== freeswitch@Linux61> jsrun alarm.js API CALL [jsrun(alarm.js)] output: OK freeswitch@Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7267f936-6bfa-11de-b2fa-f3963e050c84] 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 35 (sofia/internal/1001) Ended 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ================================================== May be I'm on the wrong way, but this is my target: 1) From the external Perl freeswitch interface call a simple js 2) Make this script dial a sip/pstn phone number at the assistance location 3) Connect the just dialled assistance location to the IVR at ext 118 and let the support people hear some info about the raising fault. Thanks in advance for any suggestion. Roberto
you also have to set session.setAutoHangup(0); or it will hangup as soon as it exits the script. On Wed, Jul 8, 2009 at 3:15 PM, <paparoga@mailinator.com> wrote:
First of all I apologise for my long post.
In the meantime I reworked the simple js as suggested (adding the domain or using transfer instead of bridge) but yet the script doesn't work.
Let me add some other info.
I just cloned the '5000' demo_ivr and reworked a little to reach my target.
Next I added:
<!-- a sample IVR --> <extension name="ivr_demo_alarm"> <condition field="destination_number" expression="^118$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr_alarm"/> </condition> </extension>
into the default.xml dialplan.
Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) and than I dial '118'.
The IVR works fine.
Next from the console:
originate sofia/my.freeswitch.address/1001 118
I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr)
That is the log: ========================================================= originate sofia/10.0.0.33/1001 118
2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] API CALL [originate(sofia/10.0.0.33/1001 118)] output: +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84
freeswitch@Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->118 in context default 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 33 (sofia/internal/1001) Ended 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ====================================================
Using the js instead it looks like if the bridge commands looks for a REGISTERED user at ext. 118, and so it fails.
Using the transfer option, as suggested, the Ekiga user (1001) get to be connected to the ext 118, bur the connection drops immediately after.
==================================================== freeswitch@Linux61> jsrun alarm.js API CALL [jsrun(alarm.js)] output: OK
freeswitch@Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7267f936-6bfa-11de-b2fa-f3963e050c84] 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 35 (sofia/internal/1001) Ended 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ==================================================
May be I'm on the wrong way, but this is my target:
1) From the external Perl freeswitch interface call a simple js 2) Make this script dial a sip/pstn phone number at the assistance location 3) Connect the just dialled assistance location to the IVR at ext 118 and let the support people hear some info about the raising fault.
Thanks in advance for any suggestion.
Roberto
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (2)
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Anthony Minessale
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paparoga@mailinator.com