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FXS problem
by Luis M. Zuccolo 07 Sep '08

07 Sep '08

Hi, I'm usign an openvox card with 2 fxo modules and 1 fxs module. When I dial from fxs extension only take the first digit dialed and tried to route the call. This digit is not in dialplan, then the call is hung up. This happend only in FreeSWITCH snapshot (2008-08-31 and 2008-09-6). In release 1.0.1 work well. It's look like that don't take the option digit-timeout configured to 2000 in openzap.conf.xml Am I' missing something else? Any help about this would be appreciated. Thanks in advance. Luis Zuccolo Console logs: (FreesSWITCH release 1.0.1) 2008-09-06 18:31:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/3:1/45662530 [2f5b3615-e144-4ce0-9e75-7fd987373e3b] 2008-09-06 18:31:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing OpenZAP->45662530@default 2008-09-06 18:31:14 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/45662530 [f19ddc6a-ab75-4e5c-b881-d00918d4ef86] 2008-09-06 18:31:14 [WARNING] mod_openzap.c:1113 on_fxo_signal() Unhandled type for channel 1:1 2008-09-06 18:31:18 [NOTICE] mod_openzap.c:1097 on_fxo_signal() Channel [OpenZAP/1:1/45662530] has been answered 2008-09-06 18:31:18 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [OpenZAP/3:1/45662530] has been answered 2008-09-06 18:31:25 [NOTICE] mod_openzap.c:1218 on_fxs_signal() Hangup OpenZAP/3:1/45662530 [CS_EXECUTE] [NORMAL_CLEARING] (FreeSWITCH snapshots) 2008-09-06 18:38:04 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel OpenZAP/3:1/4 [a269cba9-9e5a-427a-8fb4-48df1ee490ad] 2008-09-06 18:38:04 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing OpenZAP->4 in context default 2008-09-06 18:38:04 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer OpenZAP/3:1/4 to enum[4@default] 2008-09-06 18:38:04 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-09-06 18:38:04 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup OpenZAP/3:1/4 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-09-06 18:38:04 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 11 (OpenZAP/3:1/4) Ended 2008-09-06 18:38:04 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel OpenZAP/3:1/4 [CS_HANGUP] My configs file are: FreeSWITCH 1.0.1 => /etc/openzap.conf/openzap.conf FreeSWITCH snapshot => ../conf/openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 [span zt] name => OpenZAP number => 3 fxs-channel => 3 conf/autoload_configs/openzap.conf.xml config: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="9"/> </settings> <analog_spans> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </analog_spans> </configuration> __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar

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openbts
by Tamas 07 Sep '08

07 Sep '08

Hi, Have you seen this? http://openbts.sourceforge.net/background.html I guess, FreeSWITCH would be better for this ;) Regards, Tamas

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Can I find out where playStream was interrupted?
by Nic Wolff 07 Sep '08

07 Sep '08

Hi, I'm writing a FS application (called from the dialplan) that needs to know, when a user stops playback of a sound, where in the sound they were. In Asterisk, I'd pass "STREAM FILE" back from an AGI and when the user hit a key Asterisk would give me the endpos in the status line. Is there some way to get this information back from Freeswitch, either in the inputCallback function or from the session? Thanks - Nic

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detecting unresponsive gateways in javascript
by msp 06 Sep '08

06 Sep '08

when i call outbound gateway, the gateway which i call is down, so there is no response, I am running the javascript for the routing outbound calls, so how can i detect that server is down after originate, so that i route can call to another gateway Thanks msp

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IVR Timeout/Invalid Option Actions
by Marc Lewis 06 Sep '08

06 Sep '08

Is there a way to have an IVR menu take an action besides disconnecting if the digit timeout is reached or if there is an invalid option? An IVR that looks like this would be ideal, with two additional actions menu-timeout and menu-invalid. This would give the IVR a great deal more flexibility. <menu name="demo_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout ="10000" inter-digit-timeout="2000" max-failures="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/> <entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/>  <entry action="menu-sub" digits="4" param="demo_ivr_submenu"/>  <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>  <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML default"/> <entry action="menu-top" digits="9"/>  <entry action="menu-timeout" param="menu-exec-app;param=transfer 100 XML default"/>  <entry action="menu-invalid" param="menu-top"/>  </menu>  <menu name="demo_ivr_submenu" greet-long="phrase:demo_ivr_sub_menu" greet-short="phrase:demo_ivr_sub_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout="15000" max-failures="3"> <entry action="menu-top" digits="*"/> </menu> -- Marc Lewis Avvatel Corporation

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post paid account creation
by xbipin 05 Sep '08

05 Sep '08

the TLS issue on windows seems to be unsolvable by me and the interest got over when i came to know my linksys device doesnt do the TLS which freeswitch supports. wouldnt it be better if FS could be backward compatible with the TLS dont by other devices as i cant seem to find a update for the firmware also and linksys not developing also so i started searching for other stuff and came across openSBC which does a form of encryption called XOR, very basic but the good news is it bypasses most blocks as it seem to function similar to voipswitch voip tunnel. neways, i think im gonna use that as a SBC as it bypasses my block so i can now access my voip server, bytheway they also have opensipstack which supports almost all the codecs, lets me not go into it. basically what im looking for is a way directly by modifying the xml files in FS to support account balances for all configured accounts on it. i mean, that user accounts i create, can they be configured to work on credit basis or lets just make it even simpler, can i make changes to xml files so that a specific user account can make calls of for eg: 200mins to any number? after each call that number should decrease based on call duration. not using a billing software but for temporary means is it possible and if so then how? -- View this message in context: http://www.nabble.com/post-paid-account-creation-tp19339543p19339543.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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compile error (on windows with VS 2005)
by Dave 05 Sep '08

05 Sep '08

I got the compile error on windows XP pro (with VS 2005). ..\..\src\switch_xml.c(2234) : error C2220: warning treated as error - no 'object' file generated ..\..\src\switch_xml.c(2234) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2266) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2507) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2522) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2529) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2600) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data Generating Code... Results Build log was saved at "file://g:\opensource\tele\freeswitch\freeswitch-1.0.1\w32\Library\Debug\BuildLog FreeSwitchCoreLib.htm" FreeSwitchCoreLib - 1 error(s), 6 warning(s) Please help me out. ___________________________________________________________ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com.

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call through service
by Gayatri Kulkarni 05 Sep '08

05 Sep '08

Does Freeswitch implement the call through service? -- I couldnt find any relevant application. By call through service I mean: enables authorized corporate users outside CallThrough services are the most common form of service. They require no registration or pre-payment. On your existing telephone, you simply dial the service provider's number for the destination you require. You usually hear a introductory pre-recorded message and you then dial the number you want to talk to. You are charged by your existing telephone provider for the call to the initial number at the rate published by the CallThrough provider. Does Freeswitch implement the call through service? If it does, how can i invoke it through the console? Thanks a million for answering this, in advance, Regards, Gayatri Kulkarni

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Register cell phone
by Gayatri Kulkarni 05 Sep '08

05 Sep '08

Hi, Can I register my cell phone to FS? How can I do that? Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect.

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simulate att_xfer , threeway
by Lee JJ 05 Sep '08

05 Sep '08

Hello : While the att_xfer , I collect show the calls and channels info . After press "0" , the calls is Zero , and found the last channel using application "three_way" # during att_xfer , press "0" freeswitch@TTN_Li> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05 11:00:40,1220583640,sofia/inter2/1529@10.20.1.233,CS_EXECUTE,1529,1529,172.16.93.231,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05 11:00:53,1220583653,sofia/inter2/1528,CS_EXECUTE,Extension 1529,1529,172.16.93.231,1528,three_way,33c232f5-e0ff-446e-ad93-8c2f80d61656,XML,inter2,PCMU,8000,PCMU,8000 3 total. freeswitch@TTN_Li> freeswitch@TTN_Li> show calls API CALL [show(calls)] output: 0 total. freeswitch@TTN_Li> show calls API CALL [show(calls)] output: function,created,created_epoch,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2008-09-05 11:00:42,1220583642,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/1529@10.20.1.233,33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension 1529,1529,1526,sofia/inter2/1526,0d51ddb6-cac2-428b-a310-d9c54f979efc 2008-09-05 11:00:56,1220583656,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/1529@10.20.1.233,33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension 1529,1529,1528,sofia/inter2/1528,501e6351-6b9f-46a6-81bc-9a2c5a3306fa 2 total. freeswitch@TTN_Li> freeswitch@TTN_Li> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05 11:00:40,1220583640,sofia/inter2/1529@10.20.1.233,CS_EXECUTE,1529,1529,172.16.93.231,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05 11:00:53,1220583653,sofia/inter2/1528,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1528,,,XML,inter2,PCMU,8000,PCMU,8000 3 total. So , what's the no calls changing meaning ? And I simulate the three_way working using scripts, pitty thing is either hanup immediately or someone in hold music. PLS give some advice . ### call test scripts staring n_sess = new Session() ; res = n_sess.originate(n_sess, "sofia/inter2/1528%10.20.1.233" ); // wait for it ... answer n_sess.waitForAnswer(10000); if ( n_sess.ready()) { //never go into here console_log("info", "n_sess uuid: "+ n_sess.uuid + "\n"); n_sess.execute("three_way", session.uuid ) ; }

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