December 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mod_java - usage question
by Anthony Minessale 11 Dec '07

11 Dec '07

>From xml_rpc, event_socket or the console you can execute the command uuid_bridge <a_uuid> <b_uuid> and yes park will block. The correct approach would be to have an extension in your dialplan so all inbound calls execute the "park" app then from event socket or a C event consumer wait for the CHANNEL_PARK event and control the channel externally. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: jonathan augenstine <jaugenstine(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, December 11, 2007 9:49:50 AM Subject: Re: [Freeswitch-users] mod_java - usage question Anthony, Thank you for the clarification. I am beginning to understand basic behavior and design. I am constrained by the fact that the customer wanted a Java solution. I have concluded (based on discussions with Brian West) that a good solution would be to wrap the event socket interface in Java. The customer requested that I prototype the app with mod_java first. I have concluded that the final app will probably be multiple apps using both event sockets and mod_java. Also, I was to resolve some issues during the process of trying to explain my dilemas to Damjan. I have a call flow, I just need to work out the method details. Damjan, I realized the solution to my problem was to park the first incoming call. The one last detail I have not worked out is how to bridge the second call to the parked call. I think this is the solution. Will execute("park", "") block or not?? Any hints would be nice. Thanks for all the help. Jonathan On Dec 11, 2007 6:27 AM, Anthony Minessale <anthmct(a)yahoo.com> wrote: To accomplish what you are wanting with mod_java you would need a way to invoke another thread outside of the call to assume control of the channels rather than trying to do it from the call's thread. In javascript there is a jsrun api command that will launch another script in it's own thread with no calls in it. I assume the author of mod_java could extend it to do something similar so you could run a management app that listens for events and acts accordingly. Also you could use the event_socket with a standalone java app to manage the calls from a separate entity. But by design the high level interfaces block on things like playback and bridging but allow callbacks to be connected for things like read frame and dtmf and message reception. You can always do it in C where you have full control and ability to launch threads etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: jonathan augenstine <jaugenstine(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, December 11, 2007 12:58:55 AM Subject: Re: [Freeswitch-users] mod_java - usage question Damjan, What I want to do is the following: incoming call - play a prompt - bridge incoming to endpoint 1 - endpoint 1 hangs up - park the incoming call originate call to endpoint 2 - play a prompt to endpoint 2 - bridge endpoint 2 to endpoint 3 - endpoint 3 hangs up - bridge incoming call to endpoint 2 The details on how to accomplish this with mod_java are eluding me. Every approach I have attempted I have encountered roadblocks of either hitting execute blocking what I need to do or not having the right info (what needs to be passed to originate or how to play a prompt to endpoint2 or bridge to endpoint 3). I am getting stuck on the details and cannot find the docs to clarify. Jonathan On Dec 10, 2007 10:05 PM, Damjan Jovanovic <damjan(a)ecntelecoms.com> wrote: X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached, score=- 101.233, required 6, autolearn=not spam, ALL_TRUSTED -1.80, AWL 3.17, BAYES_00 -2.60, USER_IN_WHITELIST -100.00) X-ECN Telecoms-MailScanner-From: damjan(a)ecntelecoms.com X-Spam-Status: No On Mon, 2007-12-10 at 08:53 -0800, jonathan augenstine wrote: > > > > On Mon, 2007-12-10 at 00:38 -0800, jonathan augenstine wrote: > > I am trying to put together a prototype app for Freeswitch > using the > > mod_java interface, I am hitting a deadend and I need some > guidance. > > > > What I want to do is to answer an incoming call (endpoint > 1), play a > > prompt to that call, and bridge that call to another > endpoint > > (endpoint 2). After the second endpoint (endpoint 2) > disconnects, I > > want to park that call, originate a call to another endpoint > (endpoint > > 3), play a prompt to endpoint 3, and then bridge endpoint 3 > to a new > > endpoint (endpoint 4). After endpoint 4 hangs up, I want to > bridge > > endpoint 1 and endpoint 3. Then main issue I keep running > into is > > that in the dialplan and in mod_java, calls appear to block, > so that > > it has blocked me from a solution. > > > How do calls "block"? Use a remote debugger and see where it > happens. > > If I run the following code: > > > session = new JavaSession(sessionUuid); > freeswitch.console_log("notice", "STARTED!!!\n"); > session.streamFile > ("/usr/local/freeswitch/sounds/intro.wav", 0); > session.execute("bridge", > "sofia/gateway/[gatewayname]/3237773456"); > freeswitch.console_log("notice", "COMPLETED!!!\n"); > > I do not see the "COMPLETED!!!" console message until one of the > callers from the "bridge" hangs up. The > session.execute("bridge",.....) call blocks. session.execute() is supposed to block until the application finishes running. I'm not sure why that's a problem, you only want to park the call after bridge is finished. > > > My first question is, can you do this with the mod_java > interface? If > > so, how? Secondly, if it is not possible then do you need > to use the > > event_socket interface to accomplish this app? Thank you. > > > When there is no other function that does the job, try the > execute() > method in mod_java. Event sockets can probably do the same, > but I've > found them harder to work with (don't know if there are any > APIs that > make it easier), and since they cost 2 sockets and probably 1 > thread per > call and if on the same machine double the number of context > switches, > they are more resource intensive too. > > As illustrated above I am using the execute. > > > > > > Best Regards, > > Jonathan > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > Damjan > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Damjan _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-users] mod_java - usage question
by Anthony Minessale 11 Dec '07

11 Dec '07

To accomplish what you are wanting with mod_java you would need a way to invoke another thread outside of the call to assume control of the channels rather than trying to do it from the call's thread. In javascript there is a jsrun api command that will launch another script in it's own thread with no calls in it. I assume the author of mod_java could extend it to do something similar so you could run a management app that listens for events and acts accordingly. Also you could use the event_socket with a standalone java app to manage the calls from a separate entity. But by design the high level interfaces block on things like playback and bridging but allow callbacks to be connected for things like read frame and dtmf and message reception. You can always do it in C where you have full control and ability to launch threads etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: jonathan augenstine <jaugenstine(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, December 11, 2007 12:58:55 AM Subject: Re: [Freeswitch-users] mod_java - usage question Damjan, What I want to do is the following: incoming call - play a prompt - bridge incoming to endpoint 1 - endpoint 1 hangs up - park the incoming call originate call to endpoint 2 - play a prompt to endpoint 2 - bridge endpoint 2 to endpoint 3 - endpoint 3 hangs up - bridge incoming call to endpoint 2 The details on how to accomplish this with mod_java are eluding me. Every approach I have attempted I have encountered roadblocks of either hitting execute blocking what I need to do or not having the right info (what needs to be passed to originate or how to play a prompt to endpoint2 or bridge to endpoint 3). I am getting stuck on the details and cannot find the docs to clarify. Jonathan On Dec 10, 2007 10:05 PM, Damjan Jovanovic <damjan(a)ecntelecoms.com> wrote: X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached, score=- 101.233, required 6, autolearn=not spam, ALL_TRUSTED -1.80, AWL 3.17, BAYES_00 -2.60, USER_IN_WHITELIST -100.00) X-ECN Telecoms-MailScanner-From: damjan(a)ecntelecoms.com X-Spam-Status: No On Mon, 2007-12-10 at 08:53 -0800, jonathan augenstine wrote: > > > > On Mon, 2007-12-10 at 00:38 -0800, jonathan augenstine wrote: > > I am trying to put together a prototype app for Freeswitch > using the > > mod_java interface, I am hitting a deadend and I need some > guidance. > > > > What I want to do is to answer an incoming call (endpoint > 1), play a > > prompt to that call, and bridge that call to another > endpoint > > (endpoint 2). After the second endpoint (endpoint 2) > disconnects, I > > want to park that call, originate a call to another endpoint > (endpoint > > 3), play a prompt to endpoint 3, and then bridge endpoint 3 > to a new > > endpoint (endpoint 4). After endpoint 4 hangs up, I want to > bridge > > endpoint 1 and endpoint 3. Then main issue I keep running > into is > > that in the dialplan and in mod_java, calls appear to block, > so that > > it has blocked me from a solution. > > > How do calls "block"? Use a remote debugger and see where it > happens. > > If I run the following code: > > > session = new JavaSession(sessionUuid); > freeswitch.console_log("notice", "STARTED!!!\n"); > session.streamFile > ("/usr/local/freeswitch/sounds/intro.wav", 0); > session.execute("bridge", > "sofia/gateway/[gatewayname]/3237773456"); > freeswitch.console_log("notice", "COMPLETED!!!\n"); > > I do not see the "COMPLETED!!!" console message until one of the > callers from the "bridge" hangs up. The > session.execute("bridge",.....) call blocks. session.execute() is supposed to block until the application finishes running. I'm not sure why that's a problem, you only want to park the call after bridge is finished. > > > My first question is, can you do this with the mod_java > interface? If > > so, how? Secondly, if it is not possible then do you need > to use the > > event_socket interface to accomplish this app? Thank you. > > > When there is no other function that does the job, try the > execute() > method in mod_java. Event sockets can probably do the same, > but I've > found them harder to work with (don't know if there are any > APIs that > make it easier), and since they cost 2 sockets and probably 1 > thread per > call and if on the same machine double the number of context > switches, > they are more resource intensive too. > > As illustrated above I am using the execute. > > > > > > Best Regards, > > Jonathan > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > Damjan > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Damjan _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? 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mod_java - usage question
by jonathan augenstine 11 Dec '07

11 Dec '07

I am trying to put together a prototype app for Freeswitch using the mod_java interface, I am hitting a deadend and I need some guidance. What I want to do is to answer an incoming call (endpoint 1), play a prompt to that call, and bridge that call to another endpoint (endpoint 2). After the second endpoint (endpoint 2) disconnects, I want to park that call, originate a call to another endpoint (endpoint 3), play a prompt to endpoint 3, and then bridge endpoint 3 to a new endpoint (endpoint 4). After endpoint 4 hangs up, I want to bridge endpoint 1 and endpoint 3. Then main issue I keep running into is that in the dialplan and in mod_java, calls appear to block, so that it has blocked me from a solution. My first question is, can you do this with the mod_java interface? If so, how? Secondly, if it is not possible then do you need to use the event_socket interface to accomplish this app? Thank you. Best Regards, Jonathan

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conference voice path issue
by jonathan augenstine 10 Dec '07

10 Dec '07

I am working through a conference module problem I am having. I have a simple dialplan implementation that answers an incoming call, plays a prompt, and then connects the call to a conference. Initially, I would dial in on two clients (one PSTN/SIP gateway and the other a SIP client on my laptop). The freeswitch console messages indicate the calls would connect to the conference but I heard no voice path. I have been working on it this morning and simplified my dialplan to the following. I have not modified the conference.conf.xml and I am using the default values. What I am experiencing now is a tremendous amount of echo. Any pointers on what the issue might be? Jonathan Here is my dialplan: <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="/usr/local/freeswitch/sounds/PlumbleIntro.wav"/> <action application="conference" data="airespring"/> Here is the freeswitch output: call number 1: 2007-12-10 11:40:38 [NOTICE] switch_core_state_machine.c:135 switch_core_standard_on_execute() Execute conference(airespring) 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'mute' bound to '0'. 2007-12-10 11:40:38 [DEBUG] switch_ivr.c:1095 switch_ivr_digit_stream_parser_set_event() max len 1 2007-12-10 11:40:38 [DEBUG] switch_ivr.c:1099 switch_ivr_digit_stream_parser_set_event() min len 1 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'deaf mute' bound to '*'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'energy up' bound to '9'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'energy equ' bound to '8'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'energy dn' bound to '7'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol talk up' bound to '3'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol talk zero' bound to '2'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol talk dn' bound to '1'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol listen up' bound to '6'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol listen zero' bound to '5'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'vol listen dn' bound to '4'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4439 conf_default_controls() Installing default caller control action 'hangup' bound to '#'. 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4175 conference_function() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-12-10 11:40:38 [DEBUG] mod_conference.c:4214 conference_function() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-12-10 11:40:38 [DEBUG] mod_sofia.c:810 sofia_receive_message() De-activate timed RTP! 2007-12-10 11:40:38 [DEBUG] switch_core_session.c:339 switch_core_session_receive_message() Kill sofia/default/17476338055@proxy01.sipphone.com:5060 [BREAK] 2007-12-10 11:40:38 [DEBUG] mod_conference.c:1531 conference_loop_output() setup timer soft success interval: 20 samples: 160 2007-12-10 11:40:38 [DEBUG] mod_conference.c:724 conference_thread_run() setup timer success interval: 20 samples: 160 2007-12-10 11:40:38 [DEBUG] switch_core_io.c:292 switch_core_session_read_frame() Engaging Read Buffer at 320 bytes call number 2: 2007-12-10 11:44:00 [NOTICE] switch_core_state_machine.c:135 switch_core_standard_on_execute() Execute conference(airespring) 2007-12-10 11:44:00 [DEBUG] mod_conference.c:4175 conference_function() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-12-10 11:44:00 [DEBUG] mod_conference.c:4214 conference_function() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-12-10 11:44:00 [DEBUG] mod_sofia.c:810 sofia_receive_message() De-activate timed RTP! 2007-12-10 11:44:00 [DEBUG] switch_core_session.c:339 switch_core_session_receive_message() Kill sofia/default/+16266880158@72.37.161.251:5060 [BREAK] 2007-12-10 11:44:00 [DEBUG] mod_conference.c:1531 conference_loop_output() setup timer soft success interval: 20 samples: 160 2007-12-10 11:44:00 [DEBUG] switch_core_io.c:292 switch_core_session_read_frame() Engaging Read Buffer at 320 bytes

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crash stack interpretation
by Freeswitch Freeswitch 05 Dec '07

05 Dec '07

Hi, Running Freeswitch under Valgrind this morning and had a crash happen. Can someone interpret and maybe suggest where problem lies? ==4766== ==4766== Thread 19: ==4766== Invalid read of size 4 ==4766== at 0x4096743: apr_socket_sendto (sendrecv.c:119) ==4766== by 0x4045DF8: switch_socket_sendto (switch_apr.c:606) ==4766== by 0x4066F55: rtp_common_write (switch_rtp.c:1469) ==4766== by 0x4068A1B: rtp_common_read (switch_rtp.c:901) ==4766== by 0x406921C: switch_rtp_zerocopy_read_frame (switch_rtp.c:1230) ==4766== by 0x4948209: sofia_read_frame (mod_sofia.c:492) ==4766== by 0x4058113: switch_core_session_read_frame (switch_core_io.c:113) ==4766== by 0x406A736: audio_bridge_thread (switch_ivr_bridge.c:168) ==4766== by 0x406AE47: audio_bridge_on_loopback (switch_ivr_bridge.c:237) ==4766== by 0x4055F51: switch_core_session_run (switch_core_state_machine.c:409) ==4766== by 0x4053CE3: switch_core_session_thread (switch_core_session.c:681) ==4766== by 0x40989E5: dummy_worker (thread.c:138) ==4766== Address 0x14 is not stack'd, malloc'd or (recently) free'd ==4766== ==4766== Process terminating with default action of signal 11 (SIGSEGV): dumping core ==4766== Access not within mapped region at address 0x14 ==4766== at 0x4096743: apr_socket_sendto (sendrecv.c:119) ==4766== by 0x4045DF8: switch_socket_sendto (switch_apr.c:606) ==4766== by 0x4066F55: rtp_common_write (switch_rtp.c:1469) ==4766== by 0x4068A1B: rtp_common_read (switch_rtp.c:901) ==4766== by 0x406921C: switch_rtp_zerocopy_read_frame (switch_rtp.c:1230) ==4766== by 0x4948209: sofia_read_frame (mod_sofia.c:492) ==4766== by 0x4058113: switch_core_session_read_frame (switch_core_io.c:113) ==4766== by 0x406A736: audio_bridge_thread (switch_ivr_bridge.c:168) ==4766== by 0x406AE47: audio_bridge_on_loopback (switch_ivr_bridge.c:237) ==4766== by 0x4055F51: switch_core_session_run (switch_core_state_machine.c:409) ==4766== by 0x4053CE3: switch_core_session_thread (switch_core_session.c:681) ==4766== by 0x40989E5: dummy_worker (thread.c:138) --T

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default_context changed?
by Ivan C Myrvold 03 Dec '07

03 Dec '07

I have been running Freeswitch since August 9 without doing any upgrade, as I have been satisfied with how it worked for me. But I thought I should upgrade today, and I checked out the newest svn version. Running Freeswitch now, I only gets 2 of my 3 extensions authenticated with my SIP provider. 37397855 is the only one getting registered, 37048300 and 37048867 is not getting registered. I can see from the log that Freeswitch tries to authenticate 37048300 and 37048867 with 'Digest:"213.160.242.135":37397855:xxxxxx' (x-ing out the password here) Here is part of the log( I wan unable to log in to pastebin, it wouldn't accept my login name and password, even as i was already logged in to freeswitch): 2007-12-03 07:43:20 [DEBUG] sofia_reg.c:774 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"213.160.242.135":37397855:xxxxxxxx'. outbound(0x3bb230): detected NAT: 192.168.207.203 != 85.165.221.194 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_i_outbound] status [101][NAT detected] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia.c:245 sofia_event_callback() nua_i_outbound: unknown event 8: 101 NAT detected 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia_reg.c:774 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"213.160.242.135":37397855:xxxxxxxx'. 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia_reg.c:774 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"213.160.242.135":37397855:xxxxxxxx'. 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_authenticate] status [904][No matching challenge] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia.c:245 sofia_event_callback() nua_r_authenticate: unknown event 47: 904 No matching challenge 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia_reg.c:774 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"213.160.242.135":37397855:xxxxxxxx'. 2007-12-03 07:43:20 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_authenticate] status [904][No matching challenge] session: n/a 2007-12-03 07:43:20 [DEBUG] sofia.c:245 sofia_event_callback() nua_r_authenticate: unknown event 47: 904 No matching challenge And here is part of sofia_conf.xml: <configuration name="sofia.conf" description="sofia Endpoint"> <profiles> <profile name="$${domain}">  <gateways> <gateway name="ip24_37048300">  <param name="username" value="37048300"/>  <param name="realm" value="213.160.242.135"/>  <param name="password" value="xxxxxxxx"/>    <param name="proxy" value="213.160.242.135"/>   </gateway> <gateway name="ip24_37048867">  <param name="username" value="37048867"/>  <param name="realm" value="213.160.242.135"/>  <param name="password" value="xxxxxxxx"/>    <param name="proxy" value="213.160.242.135"/>   </gateway> <gateway name="ip24_37397855">  <param name="username" value="37397855"/>  <param name="realm" value="213.160.242.135"/>  <param name="password" value="xxxxxxxx"/>    <param name="proxy" value="213.160.242.135"/>   </gateway> </gateways> If I delete the two last gateways, I can get 37048300 registered. What have changed here since August? Ivan

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Re: [Freeswitch-users] FreeSwitch the Right Tool?
by Anthony Minessale 03 Dec '07

03 Dec '07

It's a joke because I broke my own self-imposed restriction on new features to write these extra modules. The users are actually crying for me to release it now much more than they are crying for stability but, I will not release it until I am happy with every aspect including the initial presentation which is the primary focus of the next beta release due in the next several days. I am still making large changes in things like the default configuration and behaviors and the arrangement of the code. Once I do a formal release we will no longer change any of this which means this is my only chance. I am also up-front which is why I am simply pointing out that it was offensive rather than starting your typical flame war. Thank you for clarifying and good luck with your investigation. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Bill Binko <bill.binko(a)mapshine.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, December 3, 2007 10:52:35 AM Subject: Re: [Freeswitch-users] FreeSwitch the Right Tool? Yikes: I certainly never intended to be offensive. By cavalier, I meant dismissive in tone. It wasn't based on anything other than the note which read, "We promised to stop making new modules or features but.......we lied." Now, there are two things that could be going on: one is that your users are crying out for stability, and you are being dismissive. Otherwise, it's just a joke, and your users understand your motivation etc. All I was doing was trying to figure out which we were jumping into. As is my nature, I am generally up-front to a fault. Again, I had no intention to offend anyone, and thank you for your response. Bill Anthony Minessale wrote: Sir, I am not sure I understand your comment about my announcement of the additional features being "cavalier" My understanding of the term, which I double checked at webster.com, is that you are suggesting that I am dismissing an important matter or have some sort of snobbish attitude. Quite frankly, I had decided against adding any new features before the release and then I decided that I must stick with my original plan which was to create a voicemail module as an exercise to ensure all the components were in place for a production system. I appreciate your feedback and comments but the cavalier remark is hard to get over..... To answer your other questions that I did not find offensive: The reason we use external tools is because computer programing is complicated and takes many thousands of hours to get right. Many of the packages we rely on have their own independent mission to provide the functionality in question and it's naive and against the nature of open source to bypass these tools in favor of a home grown solution. The main application code of FreeSWITCH contains nearly 130,000 lines of code by itself and that is without counting any of the dependency libraries. A prime example is SIP, the Sofia project works day and night for years on nothing but SIP. SIP is just 1 aspect of our application and we could never match the functionality of their sip stack on our own with all the other things we have to also focus on. Our mod_sofia is about 8,000 lines of code on top of that existing sofia stack! The reason there are so many options is because of human kind. Due to interoperability, innovation and general bright ideas brought forth by our human race, we are forced to be flexible because one man's feature is another man's bug and we must allow the 2 men to co-exist by giving them opposing configurations. In terms of general usability, You will find, if you investigate closer, that the important behavior is centralized and what you may be interpreting as "many ways" to do things is simply the method of "using" these central behaviors akin to a "skin" on a gui application. Regardless of using an embedded language, a socket connection or a simple dialplan, every case results in the same core application code doing the actual work which is intentionally designed in a pyramid of complexity with the top layer being the smallest and simplest. To summarize, This is a very complicated application and it takes a lot of comprehension of telephony and multimedia to fully understand how it works. Even having the experience, one will have to investigate the software first hand to fully understand how it works. Try to remember the first time you tried photoshop! As a policy, I make no attempts to convince anyone to use my software. Please also consider sipX, CallWeaver, Bayonne or YATE or any other open source telephony application that may be available to you. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Bill Binko <bill.binko(a)mapshine.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, December 2, 2007 4:59:15 PM Subject: [Freeswitch-users] FreeSwitch the Right Tool? Hi everyone, I'm looking at FreeSwitch as a platform for an area where we're expanding our business. I was wondering if I could get your guidance on whether it's the Right Tool for what I need. I realize this is potentially a Troll, and I apologize, but this seems so far to be a small, positive group and I thought I'd just come out and ask. First some background. I have several clients who want various value-added phone services. Basically, they want IVR systems with some bells and whistles such as recording calls, integration with their websites, and "follow me" services. My company has done similar work for clients on the GIS (mapping) side, and I have personally worked with Asterisk as a true smart PBX solution, so we are considering adding this to our offerings. We have a SIP-based VOIP provider in the colocation site we use that has attractive pricing and we have good direct connectivity to them, so we would like to this to be a SIP-only solution (no TDM hardware, etc.). As we started looking at doing this with Asterisk, we found a few things. First, there are some real issues getting clear calls through Asterisk when there is no TDM hardware in place. This seems to be due to some timing issues, and seems to be worsened by some RTC changes on Linux 2.6.x. I wasn't surprised that there were some issues, but we have had a very hard time getting it to run well on a clean distro such as CentOS without custom kernel compiles, etc. Perhaps most concerning, we aren't seeing the kind of community support we found (and came to rely on) in the open-source GIS space. Posts to the asterisk-users board get lost on the way to the list, and when they do make it through, they get responses like "buy a Digium card just for timing". Well, in our 1U servers with only a PCI-Express slot, that's going to be a trick. So, I quite literally went to Google and searched for "Asterisk alternative" and FreeSwitch came up all over the place. I installed it on our server and had it running in about an hour with only minimal pain. And the sound quality (so far) has been very good when just playing recorded sounds. Here's my concerns about it so far - let me know if they're unfounded. 1) It seems to be reliant on a huge collection of external tools. I'm ok with running the FreeSwitch recommended versions of all of them, but isn't going to be a bit fragile as these tools mature separately? 2) It seems young (as a project, not the folks on it). Things like this post to the homepage worry me: http://www.freeswitch.org/node/103 as it looks a bit cavalier. 3) It seems to have more options and less guidance that I expected. This may be a first-impression thing, but there are many ways to do everything (Dial Plan, Integration languages, etc.). That's sometimes good (Perl took that approach for years) and sometimes awful (the competing opinions get in the way). We can contribute back to this project, both with development/debugging/etc. time and with documentation help, and right now it feels like the right choice. However, I thought I'd ask for some help getting past the concerns above before we jumped in. Thanks in advance for your time, Bill Binko bill.binko(a)mapshine.com Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Freeswitch-users] FreeSwitch the Right Tool?
by Anthony Minessale 03 Dec '07

03 Dec '07

Sir, I am not sure I understand your comment about my announcement of the additional features being "cavalier" My understanding of the term, which I double checked at webster.com, is that you are suggesting that I am dismissing an important matter or have some sort of snobbish attitude. Quite frankly, I had decided against adding any new features before the release and then I decided that I must stick with my original plan which was to create a voicemail module as an exercise to ensure all the components were in place for a production system. I appreciate your feedback and comments but the cavalier remark is hard to get over..... To answer your other questions that I did not find offensive: The reason we use external tools is because computer programing is complicated and takes many thousands of hours to get right. Many of the packages we rely on have their own independent mission to provide the functionality in question and it's naive and against the nature of open source to bypass these tools in favor of a home grown solution. The main application code of FreeSWITCH contains nearly 130,000 lines of code by itself and that is without counting any of the dependency libraries. A prime example is SIP, the Sofia project works day and night for years on nothing but SIP. SIP is just 1 aspect of our application and we could never match the functionality of their sip stack on our own with all the other things we have to also focus on. Our mod_sofia is about 8,000 lines of code on top of that existing sofia stack! The reason there are so many options is because of human kind. Due to interoperability, innovation and general bright ideas brought forth by our human race, we are forced to be flexible because one man's feature is another man's bug and we must allow the 2 men to co-exist by giving them opposing configurations. In terms of general usability, You will find, if you investigate closer, that the important behavior is centralized and what you may be interpreting as "many ways" to do things is simply the method of "using" these central behaviors akin to a "skin" on a gui application. Regardless of using an embedded language, a socket connection or a simple dialplan, every case results in the same core application code doing the actual work which is intentionally designed in a pyramid of complexity with the top layer being the smallest and simplest. To summarize, This is a very complicated application and it takes a lot of comprehension of telephony and multimedia to fully understand how it works. Even having the experience, one will have to investigate the software first hand to fully understand how it works. Try to remember the first time you tried photoshop! As a policy, I make no attempts to convince anyone to use my software. Please also consider sipX, CallWeaver, Bayonne or YATE or any other open source telephony application that may be available to you. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Bill Binko <bill.binko(a)mapshine.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, December 2, 2007 4:59:15 PM Subject: [Freeswitch-users] FreeSwitch the Right Tool? Hi everyone, I'm looking at FreeSwitch as a platform for an area where we're expanding our business. I was wondering if I could get your guidance on whether it's the Right Tool for what I need. I realize this is potentially a Troll, and I apologize, but this seems so far to be a small, positive group and I thought I'd just come out and ask. First some background. I have several clients who want various value-added phone services. Basically, they want IVR systems with some bells and whistles such as recording calls, integration with their websites, and "follow me" services. My company has done similar work for clients on the GIS (mapping) side, and I have personally worked with Asterisk as a true smart PBX solution, so we are considering adding this to our offerings. We have a SIP-based VOIP provider in the colocation site we use that has attractive pricing and we have good direct connectivity to them, so we would like to this to be a SIP-only solution (no TDM hardware, etc.). As we started looking at doing this with Asterisk, we found a few things. First, there are some real issues getting clear calls through Asterisk when there is no TDM hardware in place. This seems to be due to some timing issues, and seems to be worsened by some RTC changes on Linux 2.6.x. I wasn't surprised that there were some issues, but we have had a very hard time getting it to run well on a clean distro such as CentOS without custom kernel compiles, etc. Perhaps most concerning, we aren't seeing the kind of community support we found (and came to rely on) in the open-source GIS space. Posts to the asterisk-users board get lost on the way to the list, and when they do make it through, they get responses like "buy a Digium card just for timing". Well, in our 1U servers with only a PCI-Express slot, that's going to be a trick. So, I quite literally went to Google and searched for "Asterisk alternative" and FreeSwitch came up all over the place. I installed it on our server and had it running in about an hour with only minimal pain. And the sound quality (so far) has been very good when just playing recorded sounds. Here's my concerns about it so far - let me know if they're unfounded. 1) It seems to be reliant on a huge collection of external tools. I'm ok with running the FreeSwitch recommended versions of all of them, but isn't going to be a bit fragile as these tools mature separately? 2) It seems young (as a project, not the folks on it). Things like this post to the homepage worry me: http://www.freeswitch.org/node/103 as it looks a bit cavalier. 3) It seems to have more options and less guidance that I expected. This may be a first-impression thing, but there are many ways to do everything (Dial Plan, Integration languages, etc.). That's sometimes good (Perl took that approach for years) and sometimes awful (the competing opinions get in the way). We can contribute back to this project, both with development/debugging/etc. time and with documentation help, and right now it feels like the right choice. However, I thought I'd ask for some help getting past the concerns above before we jumped in. Thanks in advance for your time, Bill Binko bill.binko(a)mapshine.com ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

Woomera
by David Villasmil Govea 03 Dec '07

03 Dec '07

Hello people, Has anyone compiled woomera and been able to make calls? The signalling seems ok, but when the call is stablished woomera returns: Woomera Sending command EVENT HELLO 0 1.0 Post Increment Woomera Sending response 200 Listener enabled Woomera Sending command EVENT HELLO 0 1.0 Post Increment Woomera Sending response 100 Starting call Woomera Sending response 201 Call Started Woomera Sending command EVENT MEDIA 1 AUDIO Woomera Sending command EVENT PROCEED 1 H225 Received alerting PDU. H225 Received connect PDU. Woomera Sending command EVENT CONNECT 1 Codec Read truncated frame of raw data. Wanted 480 and got 479 Codec Write failed: Bad file descriptor LogChan Receive thread aborted (open fail) for G.711-uLaw-64k <1> H323 Could not resolve conflict, no reverse channel. LogChan Receive thread aborted (open fail) for G.711-uLaw-64k <1> Connection ip$localhost/28843 terminated. OnCleared received down non-transferred call Sending command EVENT HANGUP 1 Woomera Sending response 100 Clearing call Anyone encounter the same problem and been able to fix it? FS version is beta2 cheers -- DVG -- Imagination is more important than knowledge Albert Einstein

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FreeSwitch the Right Tool?
by Bill Binko 03 Dec '07

03 Dec '07

Hi everyone, I'm looking at FreeSwitch as a platform for an area where we're expanding our business. I was wondering if I could get your guidance on whether it's the Right Tool for what I need. I realize this is potentially a Troll, and I apologize, but this seems so far to be a small, positive group and I thought I'd just come out and ask. First some background. I have several clients who want various value-added phone services. Basically, they want IVR systems with some bells and whistles such as recording calls, integration with their websites, and "follow me" services. My company has done similar work for clients on the GIS (mapping) side, and I have personally worked with Asterisk as a true smart PBX solution, so we are considering adding this to our offerings. We have a SIP-based VOIP provider in the colocation site we use that has attractive pricing and we have good direct connectivity to them, so we would like to this to be a SIP-only solution (no TDM hardware, etc.). As we started looking at doing this with Asterisk, we found a few things. First, there are some real issues getting clear calls through Asterisk when there is no TDM hardware in place. This seems to be due to some timing issues, and seems to be worsened by some RTC changes on Linux 2.6.x. I wasn't surprised that there were some issues, but we have had a very hard time getting it to run well on a clean distro such as CentOS without custom kernel compiles, etc. Perhaps most concerning, we aren't seeing the kind of community support we found (and came to rely on) in the open-source GIS space. Posts to the asterisk-users board get lost on the way to the list, and when they do make it through, they get responses like "buy a Digium card just for timing". Well, in our 1U servers with only a PCI-Express slot, that's going to be a trick. So, I quite literally went to Google and searched for "Asterisk alternative" and FreeSwitch came up all over the place. I installed it on our server and had it running in about an hour with only minimal pain. And the sound quality (so far) has been very good when just playing recorded sounds. Here's my concerns about it so far - let me know if they're unfounded. 1) It seems to be reliant on a huge collection of external tools. I'm ok with running the FreeSwitch recommended versions of all of them, but isn't going to be a bit fragile as these tools mature separately? 2) It seems young (as a project, not the folks on it). Things like this post to the homepage worry me: http://www.freeswitch.org/node/103 as it looks a bit cavalier. 3) It seems to have more options and less guidance that I expected. This may be a first-impression thing, but there are many ways to do everything (Dial Plan, Integration languages, etc.). That's sometimes good (Perl took that approach for years) and sometimes awful (the competing opinions get in the way). We can contribute back to this project, both with development/debugging/etc. time and with documentation help, and right now it *feels* like the right choice. However, I thought I'd ask for some help getting past the concerns above before we jumped in. Thanks in advance for your time, Bill Binko bill.binko(a)mapshine.com

4 3