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Javascript Transfer IVR
by Patrick Tran 29 Aug '08

29 Aug '08

I'm trying to create an IVR that will do the following 1. Make an outbound call 2. Play a prompt 3. Make another outbound call and connect the 2 remote ends so that they can talk to each other. 4. And I'd like the IVR to record the conversation between the 2 remote ends and also time how long they talked for before hanging up. I'd also like to do all this in javascript. I tried to do something like. session.answer( ); session.streamFile("sound.wav"); session.execute("bridge", "sofia/gateway/GatewayName/#######"); but it just seems to go crazy and repeatedly call sofia/gateway/GatewayName/#######. I also tried session.execute("transfer", "######## XML default"); but nothing seems to be happening when this line is executed. Can anyone point me in the right direction of what I'm doing wrong? -patrick

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Re: [Freeswitch-dev] Re
by Евгений Золотов 29 Aug '08

29 Aug '08

On Friday, 29 Aug 2008 14:54 Michal Bielicki wrote: > jsrun is for console, not for dialplan. The syntax of the console is > not the same as the syntax for the dialplan Whether is it the dialplan applications, if we can see such message: > show application

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Need some info about FreeSWITCH
by Евгений Золотов 29 Aug '08

29 Aug '08

We try to write some script extensions for application LUA & JS & Perl. There are some questions : 1. Seems, that for use LUA there is no necessity to install lua-5.1.3.tar.gz from http://www.lua.org/ftp/, also that LUA environment is built in a kernel of FreeSWITCH. Is it correct? 2. In the catalogue scripts it is placed test files: - 2004.pl #!/usr/bin/perl freeswitch::console_log('info', "******Perl*********\n"); my $res = `uname -a`; freeswitch::console_log('notice', "OS Info: $res\n"); 1; - 2005.lua freeswitch.consoleLog( "info", "*************LUA************\n" ); - 2006.js console_log( "*************JavaScript************\n" ); session.answer(); exit(); also we make in addition: $ luac 2005.lua -o 2005.luac and add extensions into dialplan: <extension name="2004"> <condition field="destination_number" expression="^2004$" > <action application="perl" data="2004.pl" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> <extension name="2005"> <condition field="destination_number" expression="^2005$" > <action application="lua" data="2005.lua" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> <extension name="20051"> <condition field="destination_number" expression="^20051$" > <action application="lua" data="2005.luac" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> extension name="2006"> <condition field="destination_number" expression="^2006$" > <action application="jsrun" data="2006.js" /> <action application="sleep" data="1000" /> </condition> </extension> We fulfil inquiries SIPp to the corresponding extension: $ ./sipp -sn uac -s 2004 -m 1 opos8:5070 And we takes on FreeSWITCH's console: 2008-08-29 12:28:07 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/sipp@192.168.2.108:5061 [7ac768e8-2673-465b-98a4-bba7f1f42f80] 2008-08-29 12:28:07 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->2004 in context public 2008-08-29 12:28:07 [INFO] switch_cpp.cpp:1050 console_log() ******Perl********* 2008-08-29 12:28:07 [NOTICE] switch_cpp.cpp:1050 console_log() OS Info: Linux opos8.altron.lan 2.6.18-53.1.19.el5 #1 SMP Wed May 7 08:20:19 EDT 2008 i686 i686 i386 GNU/Linux 2008-08-29 12:35:32 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/sipp@192.168.2.108:5061 [bc778692-309e-43a7-be3d-e4fa8465fc14] 2008-08-29 12:35:32 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->2005 in context public 2008-08-29 12:35:32 [INFO] switch_cpp.cpp:1050 console_log() *************LUA************ 2008-08-29 12:40:10 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/sipp@192.168.2.108:5061 [335f6192-fb5a-43ed-91ce-c8c40f3973b7] 2008-08-29 12:40:10 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->20051 in context public 2008-08-29 12:40:10 [INFO] switch_cpp.cpp:1050 console_log() *************LUA************ LUA script is fulfilled both in initial and in compiled sort... 2008-08-29 12:41:38 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/sipp@192.168.2.108:5061 [2feb67bb-d54b-49ba-a9f2-0d10c0fd5044] 2008-08-29 12:41:38 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->2006 in context public 2008-08-29 12:41:38 [ERR] switch_core_session.c:1034 switch_core_session_execute_application() Invalid Application jsrun - and that is a strange error, при чём, if to change in extension <action application="jsrun" data="2006.js" /> for: <action application="javascript" data="2006.js" /> that: 2008-08-29 12:48:16 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/sipp@192.168.2.108:5061 [1cc1f4d6-2986-4733-a2bf-84806ff0b681] 2008-08-29 12:48:16 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->2006 in context public 2008-08-29 12:48:16 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/nat/sipp@192.168.2.108:5061] has been answered - then error do not arise: 2008-08-29 12:50:12 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/nat/sipp@192.168.2.108:5061 Execute javascript(2006.js) 2008-08-29 12:50:12 [DEBUG] 2006.js:1 console_log() *************JavaScript*********** But FreeSWITCH tells: > help .... jsrun,jsrun <script> [additional_vars [...]],run a script

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Need more info on MWI for multi-tenants
by Stéphane Alnet 28 Aug '08

28 Aug '08

Hello, I'm having issues with multi-tenant MWI. I started with 1.0.1, debugged for a while, then noticed http://jira.freeswitch.org/browse/MODAPP-126 and upgraded to SVN but I'm still having the issue. (I did go over the diff between 1.0.1 and SVN and couldn't figure out what change was implemented for MODAPP-126.) The symptoom I see it is that (vm-)sofia-profile probably doesn't get passed to actual_sofia_presence_mwi_event_handler(). (I get "Cannot find profile <domain-name>" because it falls back to trying the domain name.) I can get rid of the error message by adding the domain name as an alias in sip_profiles/internal.xml but it doesn't address the problem (no NOTIFY goes out). I'm a little bit confused because I would have expected the MWI component to either (a) try to locate the Contact or maybe the "presence_id" (specified in "dial-string") for the specific user; or (b) use a Subscription information from the user. I've been working on this for a few hours and I'm at the point where I don't know enough of the internals of FreeSwitch to move forward. I'd appreciate it if someone could point me to a next step. Thank you! Stéphane

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Need some info about FreeSWITCH
by Евгений Золотов 28 Aug '08

28 Aug '08

I have one more question: 6. Our installation of FreeSWITCH : > FreeSWITCH Version 1.0.rc4 (exported) Started. after : > load mod_openzap ... 2008-08-28 10:39:28 [DEBUG] zap_isdn.c:783 zap_isdn_configure_span() opening d-channel #0 1:16 2008-08-28 10:39:28 [DEBUG] zap_isdn.c:783 zap_isdn_configure_span() opening d-channel #0 2:16 2008-08-28 10:39:28 [CONSOLE] switch_loadable_module.c:749 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] ... > oz list API CALL [oz(list)] output: +OK span: 1 type: isdn chan_count: 31 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options none +OK span: 2 type: isdn chan_count: 31 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options none On the console such messages are deduced from time to time (1 per minute or even rare): 2008-08-28 10:54:10 [WARNING] zap_zt.c:636 zt_next_event() Unhandled event 8 2008-08-28 10:54:10 [DEBUG] zap_isdn.c:591 process_event() EVENT [INVALID][1:16] STATE [DOWN] - in nessage Unhandled event mostly 8, occasionally 6 or 7 FreeSWITCH Version 1.0.trunk (9235) works on other side of E1channel: > FreeSWITCH Version 1.0.trunk (9235) Started. and there are no such messages on this side. Thus PRI signalization is correctly transfered on span E1. What does it mean ???

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Need some info about FreeSwitch
by Евгений Золотов 28 Aug '08

28 Aug '08

On Wednesday , 27 Aug 2008 Anthony Minessale wrote >What version are you using and on what platform Answering: 1. FreeSWITCH - Version 1.0.rc4 (exported) 2. OS - CentOS 5.0

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Need some info about FreeSWITCH
by Евгений Золотов 27 Aug '08

27 Aug '08

We have one more problem: Step1. FreeSWITCH is started on 192.168.2.108 at sip default profile /usr/local/freeswitch/conf/sip_profiles/default.xml We remove mandatory registration of UA: <param name="auth-calls" value="false"/> Step 2. Creating at /usr/local/freeswitch/conf/dialplan/extensions $ touch 2xxx.xml and adding extension: <extension name="20081"> <condition field="destination_number" expression="^20081$" > <action application="set" data="bypass_media=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="call_timeout=100" /> <action application="bridge" data="sofia/default/20081@localhost:5062" /> <action application="sleep" data="1000" /> </condition> </extension> <extension name="20082"> <condition field="destination_number" expression="^20082$" > <action application="set" data="bypass_media=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="call_timeout=100" /> <action application="bridge" data="sofia/default/20082@192.168.2.107:5062" /> <action application="sleep" data="1000" /> </condition> </extension> > reloadxml Step 3. We start two equal SIPp UA on 192.168.2.108 (with FreeSWITCH ) and on 192.168.2.107 in a server mode (waiting for SIP calls): $ ./sipp -sn uas -p 5062 We'll start calling SIPp UA ( in a client mode ) to both waiting for call UAs on 192.168.2.107: $ ./sipp -sn uac -s 20081 -m 1 -d 400 192.168.2.108 $ ./sipp -sn uac -s 20082 -m 1 -d 400 192.168.2.108 ==================================================================== $ ./sipp -sn uac -s 20081 -m 1 -d 400 192.168.2.108 Resolving remote host '192.168.2.108'... Done. ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(400 ms)/1.000s 5061 0.68 s 1 192.168.2.108:5060(UDP) Call limit reached (-m 1), 0.683 s period 1 ms scheduler resolution 0 calls (limit 30) Peak was 1 calls, after 0 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 1 0 0 100 <---------- 1 0 0 0 180 <---------- 1 0 0 0 183 <---------- 0 0 0 0 200 <---------- E-RTD1 1 0 0 0 ACK ----------> 1 0 Pause [ 400ms] 1 0 BYE ----------> 1 0 0 200 <---------- 1 0 0 0 ------------------------------ Test Terminated -------------------------------- - there is a call to UA, which located on the host where is FreeSWITCH placed and the sequence of connection and disconnection is fulfilled. $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 23.86 s 1 UDP 0 new calls during 0.862 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 7 s 0 Running, 1 Paused, 0 Woken up 0 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 1 0 0 0 <---------- 180 1 0 <---------- 200 1 0 0 ----------> ACK E-RTD1 1 0 0 0 ----------> BYE 1 0 0 0 <---------- 200 1 0 [ 4000ms] Pause 1 0 ------------------------------ Test Terminated -------------------------------- - the called side, to which FreeSWITCH makes bridge, receives all sequence of SIP messages ==================================================================== And that's 2nd experiment - the same, but the called UA (20082) placed on another host: $ ./sipp -sn uac -s 20082 -m 1 -d 400 192.168.2.108 Resolving remote host '192.168.2.108'... Done. ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(400 ms)/1.000s 5061 0.64 s 1 192.168.2.108:5060(UDP) Call limit reached (-m 1), 0.649 s period 1 ms scheduler resolution 0 calls (limit 30) Peak was 1 calls, after 0 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 1 0 0 100 <---------- 1 0 0 0 180 <---------- 1 0 0 0 183 <---------- 0 0 0 0 200 <---------- E-RTD1 1 0 0 0 ACK ----------> 1 0 Pause [ 400ms] 1 0 BYE ----------> 1 0 0 200 <---------- 1 0 0 0 - the causing side see SIP sequence from the FreeSWITCH side. It's OK. But the called side: $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 350.51 s 1 UDP 0 new calls during 0.509 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 12 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 1 0 0 0 <---------- 180 1 0 <---------- 200 1 9 1 ----------> ACK E-RTD1 0 0 0 0 ----------> BYE 0 0 0 0 <---------- 200 0 0 [ 4000ms] Pause 0 0 ------------------------------ Test Terminated -------------------------------- 2008-08-27 15:45:48:982 1219841148.982687: Aborting call on UDP retransmission timeout for Call-ID 'd7bd5c78-eed8-122b-8895-00d0b7169f0d'. - the called side repeats answer 200 (njrmal end of INVITE), but cann't receive ACK for it. And here UA expecting on the same UDP port, but on one host with FreeSWITCH ( i.e. another UA): $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 331.42 s 1 UDP 0 new calls during 0.412 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 6 s 0 Running, 0 Paused, 0 Woken up 20 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 0 0 0 1 <---------- 180 0 0 <---------- 200 0 0 0 ----------> ACK E-RTD1 0 0 0 0 ----------> BYE 0 0 0 0 <---------- 200 0 0 [ 4000ms] Pause 0 0 ------------------------------ Test Terminated -------------------------------- - it receives Unexpected-Msg, and that's Unexpected-Msg BYE: 2008-08-27 15:45:49:349 1219841149.349801: Dead call d7bd5c78-eed8-122b-8895-00d0b7169f0d (aborted at index 0), received 'BYE sip:127.0.0.1:5062;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.2.105;rport;branch=z9hG4bKpgDZFDN8N99gF Max-Forwards: 70 From: "sipp" <sip:sipp@192.168.2.105>;tag=pavp9yj1NmH7a To: <sip:20082@192.168.2.107:5062>;tag=24418SIPpTag011 Call-ID: d7bd5c78-eed8-122b-8895-00d0b7169f0d CSeq: 103802223 BYE Contact: <sip:mod_sofia@192.168.2.105:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.rc4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 What do we make wrong? Also do we misunderstand in configuration FreeSWITCH.

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Re: [Freeswitch-dev] Need some info about FreeSwitch
by Евгений Золотов 27 Aug '08

27 Aug '08

Thanks for all who has given answers. But I still have questions. 3. After many restarts FreeSWITCH installation it has stop to work correctly ( Version 1.0 trunk (9235) ) 2008-08-27 10:47:36 [CRIT] sofia.c:525 sofia_profile_thread_run() Cannot Open SQL Database [internal]! 2008-08-27 10:47:36 [ERR] switch_core_db.c:190 switch_core_db_open_file() SQL ERR [unable to open database file] 2008-08-27 10:47:36 [CRIT] sofia.c:525 sofia_profile_thread_run() Cannot Open SQL Database [nat]! (core dumped) We can restore it only after renaming /usr/local/freeswitch and reinstalling it again. Then we restore directoty .../conf from previous installation. It's very laborious. Does anybody know easier method of restoring (davaged databases SQLite) without installation? 4. With which application and API is possible to keep full content of RTP audiostream? Our FreeSWITCH is configured in such manner that RTP-traffic directing throw FreeSWITCH, but not between UAs. 5. As UAs we can use now only soft-clients and we use SIPp. Whether it is possible to use softsip-cli from Nokia ( that is the part of sofia library, which uses FreeSWITCH ) for our experiments?

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hangup_after_bridge and intercept application
by Francisco de Ezcurra 27 Aug '08

27 Aug '08

Hi all I'm trying to transfer the a channel after the other leg hangup in a bridge created with the application 'intercept' using the flag hangup_after_bridge, but it seems not to be working. (Below are the steps I followed). Am i doing anything wrong ? Thanks in advance Panchi -------------------------------------------------------------------------------------- 1) Make a call via mod_event_socket to a registered phone (here user2.fezcurra) and transfer the it to the extension 2100 (waiting) to wait for the other party. api originate {hangup_after_bridge=false}sofia/default/user3.fezcurra 2100 Content-Type: api/response Content-Length: 41 +OK 72a3cb6a-6943-11dd-b3aa-65ce53a0ab35 2) Set the UUID in the intercept action in the dialplan. <action application="intercept" data="72a3cb6a-6943-11dd-b3aa-65ce53a0ab35"/> 3) Make a call with another phone to the extension 2101 4) Hangup the first call originated via mod_event_socket (phone registered as user2.fezcurra). Problem: I expected the second phone to be transferred to the extension 2102 and hear the audio and then hangup, but i never hear the audio and the channel is hungup when the bridge terminates. -------------------------------------------------------------------------------------- Here are the extensions in the dialplan: <extension name="waiting"> <condition field="destination_number" expression="2100"> <action application="fifo" data="myqueue in undef test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/> </condition> </extension> <extension name="intercept"> <condition field="destination_number" expression="2101"> <action application="answer" /> <action application="set" data="hangup_after_bridge=false"/>  <action application="intercept" data="72a3cb6a-6943-11dd-b3aa-65ce53a0ab35"/> <action application="transfer" data="2102"/> </condition> </extension> <extension name="hold_music"> <condition field="destination_number" expression="2102"> <action application="playback" data="test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/> <action application="hangup" /> </condition> </extension>

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Need some info about FreeSwitch
by Евгений Золотов 26 Aug '08

26 Aug '08

We had been developed special PBX for Windows since a few years for our partners. Now we deicide to use FreeSwitch. We had been tested him for a few months, but we get some problems and questions: 1. How do we can realize SS7, probably using libraries of another developers? We need to do this for integrating third-part SS7 solutions into FreeSwitch. We have qualified specialists and we are ready let have results of our job for FreeSwitch Community, but we need some information about details of FreeSwitch. 2. When we have been worked via mod_openzap with E1 of our card we specified call-channel with approximately such kind of record: openzap/1/1 , i.e fixed channel. How do we can specify automatic choosing of сhannel, i.e. that configuring at Asterisk with help of: g/G/r/R ? We didn't find it at FreeSwitch's descriptions.

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