From msc at freeswitch.org Tue Jan 3 00:13:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Jan 2012 13:13:06 -0800 Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Schedule Message-ID: Hello all! I wanted to send out a heads up with the tentative schedule for the upcoming FreeSWITCH community conference calls. We have some good topics scheduled for the next several weeks and I wanted to make sure everyone has ample opportunity to check their calendars and make sure they are able to join us on the days when their favorite topics are being discussed. Here are the upcoming presentations: Jan 4 - Dave Kompel et. al., discussing CDR parsing and rating Jan 11 - Moises Silva from Sangoma, discussing FreeTDM and how to configure PRI cards; also he will be presenting some new GSM goodies from Sangoma Jan 18 - Muhammed Naseer (IRC: Goni) discusses his FreeSWITCH billing application: vBilling Jan 25 - Marc Olivier Chouinard (IRC: moc) will talk about a new module he is working on: mod_voicemail_ivr Feb 1 - Robert Daniel from George Washington University will share some information on how they are using games to teach FreeSWITCH/telecommunications Feb 8 - OPEN Feb 15 - Darren Schreiber et. al. from the 2600hz project will be sharing with the community some updates on 2600hz, Whistle, and blue.box Feb 22 - OPEN We definitely look forward to all of these great presentations over the next few months. If you have a suggestion for a conference call presentation then please contact me at msc at freeswitch.org so we can make arrangements. Thanks to everyone who makes FreeSWITCH such a great community! Michael S Collins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120102/39254bff/attachment.html From justlikeef at gmail.com Tue Jan 3 02:33:53 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 2 Jan 2012 18:33:53 -0500 Subject: [Freeswitch-dev] Working on building cross platform packages Message-ID: <201201021833.54060.justlikeef@gmail.com> I am working on an RPM that will build the various pieces (modules, languages, etc.) each as a seperate packags and compile cross platform, right now on OpenSuse Build service. I have run into two issues, and before I dive deep into figuring something out, I wanted to ask for opinions/information: 1) Suse and Fedora won't allow Lame in "official" packages. How hard would it be to add a configure option or a build variable or something that would exclude the mp3 code or whatever depends on Lame? The goal being to make it avaiable in the base distribution, then have a non-open license version available on one of the "community" repositories that are not as strict. Lame is the only package I can find at this point that is specifically excluded. Others, like opal, will use FFMPEG, etc. if available, but will compile without them, also... 2) Compile checking has become more stringent in recent distributions, and I am getting overflow/underflow, buffer, etc. warning, mainly from the add on libraries (codec2, openldap2) that have been fixed since the versions that are "included". Is there a method in place to easily use system packages for those things? Thanks, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120102/c0ea64ba/attachment.html From mytemike72 at gmail.com Wed Jan 4 20:02:15 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 4 Jan 2012 18:02:15 +0100 Subject: [Freeswitch-dev] Bridge calls in mod_managed Message-ID: Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120104/58d45df0/attachment-0001.html From mytemike72 at gmail.com Wed Jan 4 20:09:46 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 4 Jan 2012 18:09:46 +0100 Subject: [Freeswitch-dev] bridging calls using mod_managed Message-ID: Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120104/890c09e7/attachment.html From mgg at giagnocavo.net Wed Jan 4 20:58:34 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 4 Jan 2012 12:58:34 -0500 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: References: Message-ID: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> Can you paste the debug log from FreeSWITCH when this happens? From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Wednesday, January 04, 2012 10:10 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] bridging calls using mod_managed Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120104/606f1f77/attachment-0001.html From mytemike72 at gmail.com Wed Jan 4 21:34:58 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 4 Jan 2012 19:34:58 +0100 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> Message-ID: Hi Michael, Here it is: 2012-01-04 19:25:09.900945 [ERR] switch_cpp.cpp:1227 =============> ...A-Leg is answered 2012-01-04 19:25:09.925947 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:09.925947 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:09.925947 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:09.925947 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:09.925947 [DEBUG] sofia.c:5494 Channel sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} entering state [completed][200] 2012-01-04 19:25:09.925947 [DEBUG] sofia.c:5494 Channel sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} entering state [ready][200] 2012-01-04 19:25:09.985953 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:10.005955 [INFO] sofia.c:819 sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} Update Callee ID to "HW-MIKE-TC01" <{MyOriginatingNumber}> 2012-01-04 19:25:10.005955 [DEBUG] sofia.c:7276 dispatched freeswitch event for INFO 2012-01-04 19:25:10.125967 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2012-01-04 19:25:10.906045 [INFO] switch_channel.c:2708 sofia/external/{MyDestinationNumber}@{MyDestinationGateway} Flipping CID from "" <{MyOutboundOriginatingNumber}> to "Outbound Call" <{MyDestinationNumber}> 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:1613 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State Change CS_SOFT_EXECUTE -> CS_HIBERNATE 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/{MyDestinationNumber}@{MyDestinationGateway} [BREAK] 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:1615 ( sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway}) State Change CS_EXECUTE -> CS_HIBERNATE 2012-01-04 19:25:10.906045 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback. 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> HandleHangUp(); 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> Finalizing call in CDR table 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:362 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) Running State Change CS_HIBERNATE 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:432 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State HIBERNATE 2012-01-04 19:25:10.906045 [DEBUG] mod_sofia.c:222 sofia/external/{MyDestinationNumber}@{MyDestinationGateway} SOFIA HIBERNATE 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:749 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State Change CS_HIBERNATE -> CS_RESET 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/{MyDestinationNumber}@{MyDestinationGateway} [BREAK] 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:432 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State HIBERNATE going to sleep 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:362 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) Running State Change CS_RESET 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:413 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State RESET 2012-01-04 19:25:10.906045 [DEBUG] mod_sofia.c:166 sofia/external/{MyDestinationNumber}@{MyDestinationGateway} SOFIA RESET 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:734 sofia/external/{MyDestinationNumber}@{MyDestinationGateway} CUSTOM RESET 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:741 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State Change CS_RESET -> CS_SOFT_EXECUTE 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/{MyDestinationNumber}@{MyDestinationGateway} [BREAK] 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:413 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State RESET going to sleep 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:362 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) Running State Change CS_SOFT_EXECUTE 2012-01-04 19:25:10.906045 [DEBUG] switch_core_state_machine.c:423 ( sofia/external/{MyDestinationNumber}@{MyDestinationGateway}) State SOFT_EXECUTE 2012-01-04 19:25:10.906045 [DEBUG] mod_sofia.c:584 SOFIA SOFT_EXECUTE 2012-01-04 19:25:10.906045 [DEBUG] switch_ivr_bridge.c:759 sofia/external/{MyDestinationNumber}@{MyDestinationGateway} CUSTOM SOFT_EXECUTE 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/{MyOriginatingNumber}@{MyOriginatingGateway} [BREAK] 2012-01-04 19:25:10.906045 [DEBUG] switch_core_session.c:791 Send signal sofia/external/{MyDestinationNumber}@{MyDestinationGateway} [BREAK] 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> apiResult=+OK 29319eca-87df-437b-8e52-92a0427f12b2 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> As long as both legs are Ready(), we wait...... 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> There is one leg no longer ready... 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> Session A NO LONGER ready... 2012-01-04 19:25:10.906045 [ERR] switch_cpp.cpp:1227 =============> Session B still ready... this is done with this piece of code: private string simpleCall(string route, string orig, int timeout) { string obCause = "ERROR"; Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("{ignore_early_media=true,origination_caller_id_number=" + orig + ",originate_timeout=20}sofia/external/"+route); if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); obCause = "SUCCESS"; Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { // Play(leg_b, "Test\\dir-no_matching_results.wav"); // Play(leg_a, "Test\\dir-no_matching_results.wav"); } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { obCause = leg_b.hangupCause(); } Debug("SimpleCall() ended with: " + obCause); return obCause; } Thanks, Mike. 2012/1/4 Michael Giagnocavo > Can you paste the debug log from FreeSWITCH when this happens?**** > > ** ** > > *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Michael Lutz > *Sent:* Wednesday, January 04, 2012 10:10 AM > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* [Freeswitch-dev] bridging calls using mod_managed**** > > ** ** > > Hi, **** > > I've been working with FS for some time now, and are progressing. I am > working on a platform (in/ivr) which is handled by .net managed code > completely. I am using just 1 application which is always launched on > incomming calls. > This application immedeately connects with my MSSQL server and checks > validity of gateways, inbound number etc. Also a cdr is written and updated > periodically while the call is active. > I have als set the HandleHangUp handler to finalize the cdr and call when > the calller disconnects. > This whole mechanism works as a charm. This application then plays > voicefiles, getdigits, and everything as I would expect. > Hoewever at some point I want to dialout to an external number and connect > these two calls together. > I do not want to leave the originating .net app as long as my incomming > caller is still online (to update the cdr and finalize the call). While > these two are connected I need to update my database periodicaly > (keepalive). > I have been looking the internet to find what I need but i can't get it to > work 100%. In fact everything works, but when I bridge the two calls > together it fails.**** > > > I use the following and this whole peace of code actually works, it dials > out, aborts when someone hangsup and when the b_leg picks up they both get > audiofile turn by turn.... It even checks outbuond status and maps this to > the original call. (so really acts like a switch).**** > > Debug("Got a route=>"+route+", Let the network play > the ringtone..."); > leg_a.Execute("ring_ready", "");**** > > Debug("Creating the new session..."); > ManagedSession leg_b = new > ManagedSession("sofia/external/"+route); // route holds > {destinationNumber}@{myExternalGateway} > if (leg_b.Ready()) > { > Debug("Session_B ready!"); > string fsuuid_b = leg_b.GetVariable("uuid");**** > > Debug("Waiting for B-Leg to answer and A-Leg not > to hangup..."); > while (!leg_b.answered() && leg_a.Ready() ) { /* > Ringing... */ }**** > > if (leg_b.answered()) > { > Debug("...B-Leg is answered"); > AnswerCall(leg_a); > while (!leg_a.answered()) { } > Debug("...A-Leg is answered");**** > > // ****** BRIDGE SHOULD BE HERE ****** **** > > Debug("As long as both legs are Ready(), we > wait......"); > while (leg_b.Ready() && leg_a.Ready()) > { > Play(leg_b, > "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > Play(leg_a, > "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > }**** > > Debug("There is one leg no longer ready..."); > if (leg_a.Ready()) { Debug("Session A still > ready..."); } else { Debug("Session A NO LONGER ready...");} > if (leg_b.Ready()) { Debug("Session B still > ready..."); } else { Debug("Session B NO LONGER ready...");} > } > else > { > Debug("A-Leg aborted while dialing..."); > }**** > > if (leg_b.Ready()) > { > Debug("B-Leg still here, so hanging up > B-Leg..."); > leg_b.Hangup("NORMAL_CALL_CLEARING"); > leg_b.destroy(); > } > } > else > { > string obCause = leg_b.hangupCause();**** > > leg_a.HangUp(obCause); > Debug("Route failed with: "+obCause); > } > return true;**** > > Of course I need to bridge the two calls together so they can speak to > each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE > HERE ****** " like this:**** > > leg_a.sleep(1000, 0); > string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} > {0}", leg_a.GetUuid(), leg_b.GetUuid())); > Debug("apiResult="+apiResult);**** > > The console log shows "+OK {guid of leg_a} **** > > but at the very same moment the incomming calls is beging hung up, (I can > see the Debug message of my HandleHangUp() at exactly the as me time as the > apiResult of the bridge. > I have checked without bridge to check if codecs are the the same and they > both are PCMU 8000. > Please any help is appreciated! I have been trying to make this work for > over a week now, but it won''t work.**** > > note. I have also tried originate with the api, but gives me the same > problem when I try to bridge these two legs together...**** > > **** > > Best Regards,**** > > Mike.**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120104/b935e0ce/attachment-0001.html From mytemike72 at gmail.com Fri Jan 6 18:56:40 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Fri, 6 Jan 2012 16:56:40 +0100 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> Message-ID: <75DE83D4-CAE8-4352-810B-0FF1BACD16CD@gmail.com> Hi, Anyone here who can help me with this problem? I got really stuck on this one.. Thanks for helping!, Mike Op 4 jan. 2012 om 18:58 heeft Michael Giagnocavo het volgende geschreven: > Can you paste the debug log from FreeSWITCH when this happens? > > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > Sent: Wednesday, January 04, 2012 10:10 AM > To: freeswitch-dev at lists.freeswitch.org > Subject: [Freeswitch-dev] bridging calls using mod_managed > > Hi, > > I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. > This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. > I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. > This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. > Hoewever at some point I want to dialout to an external number and connect these two calls together. > I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). > I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. > > I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). > Debug("Got a route=>"+route+", Let the network play the ringtone..."); > leg_a.Execute("ring_ready", ""); > > Debug("Creating the new session..."); > ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} > if (leg_b.Ready()) > { > Debug("Session_B ready!"); > string fsuuid_b = leg_b.GetVariable("uuid"); > > Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); > while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } > > if (leg_b.answered()) > { > Debug("...B-Leg is answered"); > AnswerCall(leg_a); > while (!leg_a.answered()) { } > Debug("...A-Leg is answered"); > > // ****** BRIDGE SHOULD BE HERE ****** > > Debug("As long as both legs are Ready(), we wait......"); > while (leg_b.Ready() && leg_a.Ready()) > { > Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > } > > Debug("There is one leg no longer ready..."); > if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} > if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} > } > else > { > Debug("A-Leg aborted while dialing..."); > } > > if (leg_b.Ready()) > { > Debug("B-Leg still here, so hanging up B-Leg..."); > leg_b.Hangup("NORMAL_CALL_CLEARING"); > leg_b.destroy(); > } > } > else > { > string obCause = leg_b.hangupCause(); > > leg_a.HangUp(obCause); > Debug("Route failed with: "+obCause); > } > return true; > > Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: > > leg_a.sleep(1000, 0); > string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); > Debug("apiResult="+apiResult); > > The console log shows "+OK {guid of leg_a} > > but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. > I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. > Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. > > note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... > > > Best Regards, > Mike. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120106/fe001957/attachment.html From peter.olsson at visionutveckling.se Fri Jan 6 19:45:15 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 6 Jan 2012 16:45:15 +0000 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: <75DE83D4-CAE8-4352-810B-0FF1BACD16CD@gmail.com> References: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms>, <75DE83D4-CAE8-4352-810B-0FF1BACD16CD@gmail.com> Message-ID: <1FFF97C269757C458224B7C895F35F1502BF0B@cantor.std.visionutv.se> Please pastebin some debug logs, so we know where to start looking. /Peter ________________________________ Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Michael Lutz [mytemike72 at gmail.com] Skickat: den 6 januari 2012 16:56 Till: freeswitch-dev at lists.freeswitch.org ?mne: Re: [Freeswitch-dev] bridging calls using mod_managed Hi, Anyone here who can help me with this problem? I got really stuck on this one.. Thanks for helping!, Mike Op 4 jan. 2012 om 18:58 heeft Michael Giagnocavo > het volgende geschreven: Can you paste the debug log from FreeSWITCH when this happens? From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Wednesday, January 04, 2012 10:10 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] bridging calls using mod_managed Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:4f07185e32767836217473! From mytemike72 at gmail.com Fri Jan 6 20:22:00 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Fri, 6 Jan 2012 18:22:00 +0100 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: <1FFF97C269757C458224B7C895F35F1502BF0B@cantor.std.visionutv.se> References: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> <75DE83D4-CAE8-4352-810B-0FF1BACD16CD@gmail.com> <1FFF97C269757C458224B7C895F35F1502BF0B@cantor.std.visionutv.se> Message-ID: I allready did paste logs last week on request... What logs do you want? Regards, Mike Op 6 jan. 2012 om 17:45 heeft Peter Olsson het volgende geschreven: > Please pastebin some debug logs, so we know where to start looking. > > /Peter > ________________________________ > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Michael Lutz [mytemike72 at gmail.com] > Skickat: den 6 januari 2012 16:56 > Till: freeswitch-dev at lists.freeswitch.org > ?mne: Re: [Freeswitch-dev] bridging calls using mod_managed > > Hi, > > Anyone here who can help me with this problem? I got really stuck on this one.. > > Thanks for helping!, > Mike > > Op 4 jan. 2012 om 18:58 heeft Michael Giagnocavo > het volgende geschreven: > > Can you paste the debug log from FreeSWITCH when this happens? > > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > Sent: Wednesday, January 04, 2012 10:10 AM > To: freeswitch-dev at lists.freeswitch.org > Subject: [Freeswitch-dev] bridging calls using mod_managed > > > Hi, > I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. > This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. > I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. > This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. > Hoewever at some point I want to dialout to an external number and connect these two calls together. > I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). > I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. > > I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). > > Debug("Got a route=>"+route+", Let the network play the ringtone..."); > leg_a.Execute("ring_ready", ""); > > Debug("Creating the new session..."); > ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} > if (leg_b.Ready()) > { > Debug("Session_B ready!"); > string fsuuid_b = leg_b.GetVariable("uuid"); > > Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); > while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } > > if (leg_b.answered()) > { > Debug("...B-Leg is answered"); > AnswerCall(leg_a); > while (!leg_a.answered()) { } > Debug("...A-Leg is answered"); > > // ****** BRIDGE SHOULD BE HERE ****** > > Debug("As long as both legs are Ready(), we wait......"); > while (leg_b.Ready() && leg_a.Ready()) > { > Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > } > > Debug("There is one leg no longer ready..."); > if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} > if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} > } > else > { > Debug("A-Leg aborted while dialing..."); > } > > if (leg_b.Ready()) > { > Debug("B-Leg still here, so hanging up B-Leg..."); > leg_b.Hangup("NORMAL_CALL_CLEARING"); > leg_b.destroy(); > } > } > else > { > string obCause = leg_b.hangupCause(); > > leg_a.HangUp(obCause); > Debug("Route failed with: "+obCause); > } > return true; > > Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: > > leg_a.sleep(1000, 0); > string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); > Debug("apiResult="+apiResult); > > The console log shows "+OK {guid of leg_a} > > but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. > I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. > Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. > > note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... > > Best Regards, > > Mike. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > !DSPAM:4f07185e32767836217473! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From khovayko at gmail.com Sat Jan 7 23:29:03 2012 From: khovayko at gmail.com (Oleg Khovayko) Date: Sat, 07 Jan 2012 15:29:03 -0500 Subject: [Freeswitch-dev] FreeBSD build problem: no tz_t Message-ID: <4F08AB0F.2080609@gmail.com> Hi, I tried pull recent GIT copy, and build by bootstrap.sh + make all, on the FreeBSD v 7.2. *See error:* In file included from /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' gmake[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 gmake: *** [all] Error 2 *I assume, tz_t is not defined:* /*! Set the time zone for the time stamp in page header lines. If this function is not used the current time zone of the program's environment is used. \brief Set the header timezone. \param s The T.4 context. \param tz A time zone descriptor. */ SPAN_DECLARE(void) t4_tx_set_header_tz(t4_state_t *s, tz_t *tz); # man tz_t No manual entry for tz_t *How to define it correctly?* Thanks, Oleg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120107/d411471a/attachment.html From peter.olsson at visionutveckling.se Sun Jan 8 00:03:42 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 7 Jan 2012 21:03:42 +0000 Subject: [Freeswitch-dev] FreeBSD build problem: no tz_t In-Reply-To: <4F08AB0F.2080609@gmail.com> References: <4F08AB0F.2080609@gmail.com> Message-ID: <1FFF97C269757C458224B7C895F35F1502C146@cantor.std.visionutv.se> Remove "libs/spandsp/src/spandsp.h", then issue a "make spandsp-reconf", and then rebuild. It's been discussed multiple times on the mailing list. /Peter ________________________________ Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Oleg Khovayko [khovayko at gmail.com] Skickat: den 7 januari 2012 21:29 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] FreeBSD build problem: no tz_t Hi, I tried pull recent GIT copy, and build by bootstrap.sh + make all, on the FreeBSD v 7.2. See error: In file included from /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' gmake[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 gmake: *** [all] Error 2 I assume, tz_t is not defined: /*! Set the time zone for the time stamp in page header lines. If this function is not used the current time zone of the program's environment is used. \brief Set the header timezone. \param s The T.4 context. \param tz A time zone descriptor. */ SPAN_DECLARE(void) t4_tx_set_header_tz(t4_state_t *s, tz_t *tz); # man tz_t No manual entry for tz_t How to define it correctly? Thanks, Oleg !DSPAM:4f08a9dc32763461937777! From juanito1982 at gmail.com Sun Jan 8 15:40:55 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sun, 8 Jan 2012 13:40:55 +0100 Subject: [Freeswitch-dev] Answered channel Message-ID: Hello, Is there any way to know if a channel has been answered having a reference to that channel? Channel profile answered time seems to be setted up as soon early media is received. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120108/51681996/attachment.html From juanito1982 at gmail.com Sun Jan 8 16:02:43 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sun, 8 Jan 2012 14:02:43 +0100 Subject: [Freeswitch-dev] Answered channel In-Reply-To: References: Message-ID: Sorry, I made a mistake. All is working OK. Regards 2012/1/8 Juan Antonio Iba?ez Santorum > Hello, > > Is there any way to know if a channel has been answered having a > reference to that channel? Channel profile answered time seems to be setted > up as soon early media is received. > > Regards > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120108/70202b7a/attachment.html From mytemike72 at gmail.com Mon Jan 9 12:47:48 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 9 Jan 2012 10:47:48 +0100 Subject: [Freeswitch-dev] bridging calls using mod_managed In-Reply-To: References: <83FF8D7C9F526E44B77C97DD2891652A12A204C2@mse17be1.mse17.exchange.ms> <75DE83D4-CAE8-4352-810B-0FF1BACD16CD@gmail.com> <1FFF97C269757C458224B7C895F35F1502BF0B@cantor.std.visionutv.se> Message-ID: Hi, I did managed to get something to work. I placed complete code on pastebin: http://pastebin.com/eCsYSdfk I don't know why exactly it works now, I think it has something to do with a public decalration of my leg_a as managed session and using that, now I pass the contect.Session every time and create that as a local leg_a reference and that seems to work... Still I have some problems with I need realy need somebody to help me with... 1. immediately after the bridge (log says exact same timestamp) the HangUp handler is invoked. (it seems as a separate part, as the rest of my code is handled too) Why is the hangup handler called when the bridge is started and how to avoid this? 2. After the bridge I need to get into a loop to continue my code while in the bridge, and after the bridge, depending on who hung up. So after bridge I go into while loop like: while(leg_a.bridged() && leg_b.bridged()) { } But the both return false immediately. Al other functions ( contect.Session.Ready() ) return false too, though my code continues and just continues my function. How can I make this work? My (custom debug)log shows: 2012-01-09 10:29:42.111746 [INFO] sofia.c:819 sofia/internal/****@*.*.*.*Update Callee ID to "HW-MIKE-TC01" <3120*******> 2012-01-09 10:29:42.993834 [INFO] switch_channel.c:2708 sofia/external/***********@*.*.*.* Flipping CID from "" <313*********> to "Outbound Call" <316*********> 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> HandleHangUp(); 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> Finalizing call in CDR table 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> apiResult=+OK 792d63a7-fa2e-43fb-b43b-32c11e2b7452 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> As long as both legs are Ready(), we wait...... 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> There is one leg no longer ready... 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> Session A NO LONGER ready... 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> Session B NO LONGER ready... 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> SimpleCall() ended with: SUCCESS 2012-01-09 10:29:42.993834 [ERR] switch_cpp.cpp:1227 =============> End of run... Then when I realy hangup: 2012-01-09 10:31:13.386873 [NOTICE] sofia.c:623 Hangup sofia/external/316********@*.*.*.* [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2012-01-09 10:31:13.406875 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/internal/3120********@*.*.*.* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2012-01-09 10:31:13.406875 [NOTICE] switch_core_session.c:1398 Session 133 ( sofia/internal/3120*******@*.*.*.*) Ended 2012-01-09 10:31:13.406875 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/3120*******@*.*.*.* [CS_DESTROY] Thanks in advance! Regards, Mike. 2012/1/6 Michael Lutz > I allready did paste logs last week on request... What logs do you want? > > Regards, > Mike > > > Op 6 jan. 2012 om 17:45 heeft Peter Olsson < > peter.olsson at visionutveckling.se> het volgende geschreven: > > > Please pastebin some debug logs, so we know where to start looking. > > > > /Peter > > ________________________________ > > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [ > freeswitch-dev-bounces at lists.freeswitch.org] f?r Michael Lutz [ > mytemike72 at gmail.com] > > Skickat: den 6 januari 2012 16:56 > > Till: freeswitch-dev at lists.freeswitch.org > > ?mne: Re: [Freeswitch-dev] bridging calls using mod_managed > > > > Hi, > > > > Anyone here who can help me with this problem? I got really stuck on > this one.. > > > > Thanks for helping!, > > Mike > > > > Op 4 jan. 2012 om 18:58 heeft Michael Giagnocavo > het volgende geschreven: > > > > Can you paste the debug log from FreeSWITCH when this happens? > > > > From: freeswitch-dev-bounces at lists.freeswitch.org freeswitch-dev-bounces at lists.freeswitch.org> [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > > Sent: Wednesday, January 04, 2012 10:10 AM > > To: freeswitch-dev at lists.freeswitch.org freeswitch-dev at lists.freeswitch.org> > > Subject: [Freeswitch-dev] bridging calls using mod_managed > > > > > > Hi, > > I've been working with FS for some time now, and are progressing. I am > working on a platform (in/ivr) which is handled by .net managed code > completely. I am using just 1 application which is always launched on > incomming calls. > > This application immedeately connects with my MSSQL server and checks > validity of gateways, inbound number etc. Also a cdr is written and updated > periodically while the call is active. > > I have als set the HandleHangUp handler to finalize the cdr and call > when the calller disconnects. > > This whole mechanism works as a charm. This application then plays > voicefiles, getdigits, and everything as I would expect. > > Hoewever at some point I want to dialout to an external number and > connect these two calls together. > > I do not want to leave the originating .net app as long as my incomming > caller is still online (to update the cdr and finalize the call). While > these two are connected I need to update my database periodicaly > (keepalive). > > I have been looking the internet to find what I need but i can't get it > to work 100%. In fact everything works, but when I bridge the two calls > together it fails. > > > > I use the following and this whole peace of code actually works, it > dials out, aborts when someone hangsup and when the b_leg picks up they > both get audiofile turn by turn.... It even checks outbuond status and maps > this to the original call. (so really acts like a switch). > > > > Debug("Got a route=>"+route+", Let the network play > the ringtone..."); > > leg_a.Execute("ring_ready", ""); > > > > Debug("Creating the new session..."); > > ManagedSession leg_b = new > ManagedSession("sofia/external/"+route); // route holds > {destinationNumber}@{myExternalGateway %7bmyExternalGateway>} > > if (leg_b.Ready()) > > { > > Debug("Session_B ready!"); > > string fsuuid_b = leg_b.GetVariable("uuid"); > > > > Debug("Waiting for B-Leg to answer and A-Leg not > to hangup..."); > > while (!leg_b.answered() && leg_a.Ready() ) { /* > Ringing... */ } > > > > if (leg_b.answered()) > > { > > Debug("...B-Leg is answered"); > > AnswerCall(leg_a); > > while (!leg_a.answered()) { } > > Debug("...A-Leg is answered"); > > > > // ****** BRIDGE SHOULD BE HERE ****** > > > > Debug("As long as both legs are Ready(), we > wait......"); > > while (leg_b.Ready() && leg_a.Ready()) > > { > > Play(leg_b, > "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > > Play(leg_a, > "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED > > } > > > > Debug("There is one leg no longer ready..."); > > if (leg_a.Ready()) { Debug("Session A still > ready..."); } else { Debug("Session A NO LONGER ready...");} > > if (leg_b.Ready()) { Debug("Session B still > ready..."); } else { Debug("Session B NO LONGER ready...");} > > } > > else > > { > > Debug("A-Leg aborted while dialing..."); > > } > > > > if (leg_b.Ready()) > > { > > Debug("B-Leg still here, so hanging up > B-Leg..."); > > leg_b.Hangup("NORMAL_CALL_CLEARING"); > > leg_b.destroy(); > > } > > } > > else > > { > > string obCause = leg_b.hangupCause(); > > > > leg_a.HangUp(obCause); > > Debug("Route failed with: "+obCause); > > } > > return true; > > > > Of course I need to bridge the two calls together so they can speak to > each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE > HERE ****** " like this: > > > > leg_a.sleep(1000, 0); > > string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} > {0}", leg_a.GetUuid(), leg_b.GetUuid())); > > Debug("apiResult="+apiResult); > > > > The console log shows "+OK {guid of leg_a} > > > > but at the very same moment the incomming calls is beging hung up, (I > can see the Debug message of my HandleHangUp() at exactly the as me time as > the apiResult of the bridge. > > I have checked without bridge to check if codecs are the the same and > they both are PCMU 8000. > > Please any help is appreciated! I have been trying to make this work for > over a week now, but it won''t work. > > > > note. I have also tried originate with the api, but gives me the same > problem when I try to bridge these two legs together... > > > > Best Regards, > > > > Mike. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org FreeSWITCH-dev at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > !DSPAM:4f07185e32767836217473! > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120109/5ff2f390/attachment-0001.html From Giovanni.Visciano at italtel.it Mon Jan 9 15:30:35 2012 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Mon, 9 Jan 2012 13:30:35 +0100 Subject: [Freeswitch-dev] [Minor fix] switch_core_session.c Message-ID: Minor stuff, missing comma after "T38_DESCRIPTION". switch_core_session.c, line 633 Regards, Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120109/c22caf7f/attachment.html From msc at freeswitch.org Mon Jan 9 22:48:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 11:48:35 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call This Week: Sangoma Message-ID: Greetings all! This week's conference call will feature Moises Silva from Sangoma. Moises (IRC: moy) has lots of experience with TDM communications and is the perfect person to present information on configuring PRI cards and FreeTDM. He also has some interesting information about new GSM stuff from Sangoma. We look forward to his presentation this coming Wednesday, Jan 11. The agenda page is in the usual placeon the wiki. Talk to you all on Wednesday! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120109/2422e3e0/attachment.html From khovayko at gmail.com Tue Jan 10 04:37:20 2012 From: khovayko at gmail.com (Oleg Khovayko) Date: Mon, 09 Jan 2012 20:37:20 -0500 Subject: [Freeswitch-dev] voicemail - Message is less than minimum record length Message-ID: <4F0B9650.4020705@gmail.com> Hi, I have a problem, maybe someone know, how to fix it. OS: FreeBSD v 7.2 Freeswitch: GIT master/head, 2012-01-05 I receive call from Dingaling (google voice), answer. Make bridge, and go to voicemail, if bridge was unsuccessful. Voicemail talking prompt, invite to recording... But instantly says "Message is less than minimum record length, try again". After 2-4 fails/iterations, I able to write voicemail. 2012-01-09 20:24:41.050331 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2012-01-09 20:24:41.154669 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels 20ms 2012-01-09 20:24:45.843641 [DEBUG] switch_ivr_play_say.c:1678 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2012-01-09 20:24:46.956400 [DEBUG] switch_ivr_play_say.c:585 Raw Codec Activated 2012-01-09 20:24:46.956400 [DEBUG] switch_core_codec.c:116 DingaLing/new Push codec L16:70 2012-01-09 20:24:48.963089 [DEBUG] switch_core_codec.c:141 DingaLing/new Restore previous codec PCMU:0. 2012-01-09 20:24:48.963089 [DEBUG] mod_voicemail.c:1198 Message is less than minimum record length: 3, discarding it. 2012-01-09 20:24:48.963089 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] . . . .(thereafter ~ 20 lines like following) . . . 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it From Claudio.Cavalera at italtel.it Tue Jan 10 16:41:37 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Tue, 10 Jan 2012 14:41:37 +0100 Subject: [Freeswitch-dev] Epoch current time difference In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A10198F5B@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A0FFD9930@mse17be1.mse17.exchange.ms><83FF8D7C9F526E44B77C97DD2891652A10198E17@mse17be1.mse17.exchange.ms> <83FF 8D7C9F526E44B77C97DD2891652A10198F5B@mse17be1.mse17.exchange.ms> Message-ID: Hello Michael, I've added the info on this thread to the wiki: http://wiki.freeswitch.org/wiki/Clock although poorly formatted I'm sure it will be useful to many users. However I don't understand this final question of yours (which got no answers btw). Could you please elaborate on it a bit? With "using realtime clock" do you mean "using the -nort flag"? Thanks, Claudio From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Friday, December 23, 2011 3:44 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Epoch current time difference So using realtime clock then won't screw with audio processing? -Michael Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From juraj.fabo at gmail.com Tue Jan 10 20:18:31 2012 From: juraj.fabo at gmail.com (Juraj Fabo) Date: Tue, 10 Jan 2012 18:18:31 +0100 Subject: [Freeswitch-dev] ftdm_call_place and caller_data.call_id Message-ID: Hi, When posting the original question I was working with the testapplication, rather a prototype based on the testapplications from the freeswitch git to become familiar with the freetdm library. However, now when I wanted to use this approach in application which would use only public freetdm library api and headers, I had a problem caused by fact that the ftdm_channel is declared in the include/private/ftdm_code.h header and not known to application. Please, is then my approach from the post earlier really proper way? Are the applications which load and use the freetdm supposed to know the private/core declarations? Thank you Juraj >Hello, > >On Sat, Aug 27, 2011 at 10:37 AM, Juraj Fabo > wrote: >>**>>* - from sources I see, that after the ftdm_call_place_ex(..) application could*>>* use caller_data.fchan->caller_data.call_id*>>* I tested it and it was ok, please is this a proper way how to use this API*>>* function?*>>**> > Yes that is the proper way. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120110/5a09eed2/attachment.html From huw.selley at netdev.co.uk Tue Jan 10 19:43:36 2012 From: huw.selley at netdev.co.uk (Huw Selley) Date: Tue, 10 Jan 2012 16:43:36 +0000 Subject: [Freeswitch-dev] strange module loading problem Message-ID: <96F2A94C-AF86-4E08-8C3E-4CA362194793@netdev.co.uk> Hi, I hope i have directed this question to the correct list! I have recently been upgrading a test platform to centos 6.2 (64bit) and upgrading freeswitch in the process. I use a custom module and have noticed since the upgrade that it frequently fails to load and throws this error: 2012-01-10 15:42:48.681677 [ERR] switch_xml_config.c:159 Required parameter [?a??ff.?] is missing I can restart freeswitch (via the init script) and most of the time it will then load correctly. I am fairly sure there is nothing retarded happening in the module - it's been in use for some months with no changes and I have never seen this before. The config file contents are sane which is corroborated by the fact that it sometimes works. My freeswitch is built from a recent git: FreeSWITCH Version 1.0.head (git-5b7e201 2012-01-09 17-17-03 -0600) Any ideas? I am at a loss as to how to debug this further. Sorry this post is a bit light on info but I can supply more if someone wouldn't mind directing me to what I need to supply? Thanks Huw From anthony.minessale at gmail.com Tue Jan 10 23:56:38 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Jan 2012 14:56:38 -0600 Subject: [Freeswitch-dev] strange module loading problem In-Reply-To: <96F2A94C-AF86-4E08-8C3E-4CA362194793@netdev.co.uk> References: <96F2A94C-AF86-4E08-8C3E-4CA362194793@netdev.co.uk> Message-ID: make sure your config file is properly utf-8 encoded. On Tue, Jan 10, 2012 at 10:43 AM, Huw Selley wrote: > Hi, > > I hope i have directed this question to the correct list! > I have recently been upgrading a test platform to centos 6.2 (64bit) and > upgrading freeswitch in the process. I use a custom module and have noticed > since the upgrade that it frequently fails to load and throws this error: > > 2012-01-10 15:42:48.681677 [ERR] switch_xml_config.c:159 Required > parameter [?a??ff.?] is missing > > I can restart freeswitch (via the init script) and most of the time it > will then load correctly. I am fairly sure there is nothing retarded > happening in the module - it's been in use for some months with no changes > and I have never seen this before. > The config file contents are sane which is corroborated by the fact that > it sometimes works. > > My freeswitch is built from a recent git: > > FreeSWITCH Version 1.0.head (git-5b7e201 2012-01-09 17-17-03 -0600) > > Any ideas? I am at a loss as to how to debug this further. > Sorry this post is a bit light on info but I can supply more if someone > wouldn't mind directing me to what I need to supply? > > Thanks > Huw > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120110/a8001914/attachment-0001.html From moises.silva at gmail.com Wed Jan 11 01:55:10 2012 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 10 Jan 2012 17:55:10 -0500 Subject: [Freeswitch-dev] ftdm_call_place and caller_data.call_id In-Reply-To: References: Message-ID: On Tue, Jan 10, 2012 at 12:18 PM, Juraj Fabo wrote: > Hi, > > When posting the original question I was working with the testapplication, rather a prototype based on the testapplications > from the freeswitch git to become familiar with the freetdm library. > However, now when I wanted to use this approach in application which would use only public freetdm library api and headers, > > I had a problem caused by fact that the ftdm_channel is declared in the include/private/ftdm_code.h header and not known to application. > > Please, is then my approach from the post earlier really proper way? > Are the applications which load and use the freetdm supposed to know the private/core declarations? > > You are better off using mod_freetdm.c as the reference, the sample/test applications may be outdated. ftdm_channel_t is an opaque type to API users. I don't remember what your exact question was. You should not use fchan.caller_data, you always get a copy of caller_data when you need it, the fact that is internally attached to the fchan pointer is not something you should rely on (and that's why is in a private header). *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120110/f17bbef0/attachment.html From huw.selley at netdev.co.uk Wed Jan 11 11:52:55 2012 From: huw.selley at netdev.co.uk (Huw Selley) Date: Wed, 11 Jan 2012 08:52:55 +0000 Subject: [Freeswitch-dev] strange module loading problem In-Reply-To: References: <96F2A94C-AF86-4E08-8C3E-4CA362194793@netdev.co.uk> Message-ID: <6455ABE2-F5A6-4A4E-8109-39D9400659AE@netdev.co.uk> On 10 Jan 2012, at 20:56, Anthony Minessale wrote: > make sure your config file is properly utf-8 encoded. Ahhhh that got it! Thanks :) From huw.selley at netdev.co.uk Wed Jan 11 13:48:23 2012 From: huw.selley at netdev.co.uk (Huw Selley) Date: Wed, 11 Jan 2012 10:48:23 +0000 Subject: [Freeswitch-dev] strange module loading problem In-Reply-To: References: <96F2A94C-AF86-4E08-8C3E-4CA362194793@netdev.co.uk> Message-ID: <78E68823-87DE-4F12-8338-F100EF091377@netdev.co.uk> On 10 Jan 2012, at 20:56, Anthony Minessale wrote: > make sure your config file is properly utf-8 encoded. Hmmm, I spoke too soon :( The file is UTF8 encoded (according to vim) and it's only using plain latin chars - nothing funny going on there. However I still have this problem but only sometimes. Considering that this error doesn't always happen that suggests to me that it's not the config file encoding being wrong (I assume if the file was badly encoded it would fail to load 100% of the time?) Is there anything else I can check? Thanks again Huw From msc at freeswitch.org Sat Jan 14 04:35:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 17:35:06 -0800 Subject: [Freeswitch-dev] Are you using mod_httapi Message-ID: Hello all! I know that many of you already know about mod_httapi. I am collecting examples and experiences from people who have had a chance to try it out. If you are using mod_httapi, even just for testing/tinkering, please contact me off list. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120113/4dc6d336/attachment.html From brunkb at gmail.com Sun Jan 15 19:33:51 2012 From: brunkb at gmail.com (Ben Brunk) Date: Sun, 15 Jan 2012 10:33:51 -0600 Subject: [Freeswitch-dev] build error: undefined reference to `su_log_default' in sofia-sip build Message-ID: Hi, I have the latest code I am trying to build on Ubuntu 11 and am running into a problem in freeswitch/libs/sofia-sip/ libsofia-sip-ua/su The error is: ./.libs/libsu.a(su.o): In function `su_init': su.c:(.text+0x93): undefined reference to `su_log_default' su.c:(.text+0xa2): undefined reference to `su_log_global' ./.libs/libsu.a(su_log.o): In function `su_log': su_log.c:(.text+0xfb): undefined reference to `su_log_default' ./.libs/libsu.a(su_log.o): In function `su_vllog': su_log.c:(.text+0x225): undefined reference to `su_log_default' su_log.c:(.text+0x258): undefined reference to `su_log_default' su_log.c:(.text+0x267): undefined reference to `su_log_default' ./.libs/libsu.a(su_log.o): In function `su_log_init': su_log.c:(.text+0x2dc): undefined reference to `su_log_default' ./.libs/libsu.a(su_log.o):su_log.c:(.text+0x2e9): more undefined references to `su_log_default' follow collect2: ld returned 1 exit status I have followed all of the directions for building on Ubuntu thus far and I could tell that the configuration step went smoothly. I have not undertaken any in-depth troubleshooting (yet) as I am not very experienced with the Freeswitch code base at the moment. It seems like I missed some minor configuration step or perhaps there's an OS issue going on here. Thanks for any help! Ben Brunk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120115/ed8e7e01/attachment.html From krice at freeswitch.org Sun Jan 15 21:22:21 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 15 Jan 2012 12:22:21 -0600 Subject: [Freeswitch-dev] build error: undefined reference to `su_log_default' in sofia-sip build In-Reply-To: Message-ID: Its probably an OS issue... Its is highly recommended that you run FreeSWITCH on the LTS versions of your chosen platform. For Ubuntu this is like 8.04 or 10.04. Once 12.04 is released and ready FreeSwitch will be supported there. On 1/15/12 10:33 AM, "Ben Brunk" wrote: > Hi, > > I have the latest code I am trying to build on Ubuntu 11 and am running into a > problem in freeswitch/libs/sofia-sip/ > libsofia-sip-ua/su > > The error is: > > ./.libs/libsu.a(su.o): In function `su_init': > su.c:(.text+0x93): undefined reference to `su_log_default' > su.c:(.text+0xa2): undefined reference to `su_log_global' > ./.libs/libsu.a(su_log.o): In function `su_log': > su_log.c:(.text+0xfb): undefined reference to `su_log_default' > ./.libs/libsu.a(su_log.o): In function `su_vllog': > su_log.c:(.text+0x225): undefined reference to `su_log_default' > su_log.c:(.text+0x258): undefined reference to `su_log_default' > su_log.c:(.text+0x267): undefined reference to `su_log_default' > ./.libs/libsu.a(su_log.o): In function `su_log_init': > su_log.c:(.text+0x2dc): undefined reference to `su_log_default' > ./.libs/libsu.a(su_log.o):su_log.c:(.text+0x2e9): more undefined references to > `su_log_default' follow > collect2: ld returned 1 exit status > > I have followed all of the directions for building on Ubuntu thus far and I > could tell that the configuration step went smoothly.? I have not undertaken > any in-depth troubleshooting (yet) as I am not very experienced with the > Freeswitch code base at the moment.? It seems like I missed some minor > configuration step or perhaps there's an OS issue going on here.? Thanks for > any help! > > > Ben Brunk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120115/5586f910/attachment-0001.html From msc at freeswitch.org Tue Jan 17 01:54:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 14:54:21 -0800 Subject: [Freeswitch-dev] Nous avons besoin de votre assistance avec les traductions! Message-ID: Salut! If you can read the subject line then you already know what I need: help with some French translations. As you know, our very own Marc Olivier Chouinard (Moc) has produced a set of French-Canadian sound prompts for everyone's benefit. The FreeSWITCH team and Barracuda Networks recently added some prompts to that set. We are looking to expand that sound set even more with some of our newest sound prompts. I have a grand total of about 220 prompts that I would like to get translated into French. Many of these are less-common prompts that a lot of people won't need, but I would like to get the translations done anyway so that we can get the most useful ones recorded as soon as possible. Also, some of these prompts are silly ones that may not translate too well into French. (I've heard that the French don't have a sense of humor but I cannot confirm that report at this time. :) If you'd like to help with this translation project please email me off list and I will send you a spreadsheet with what we have to do. Merci beaucoup pour tout votre soutien! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120116/5fe2a1ff/attachment.html From devel at thom.fr.eu.org Tue Jan 17 02:19:22 2012 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Tue, 17 Jan 2012 00:19:22 +0100 Subject: [Freeswitch-dev] Nous avons besoin de votre assistance avec les traductions! In-Reply-To: References: Message-ID: <002b01ccd4a5$4d0bc0d0$e7234270$@fr.eu.org> Hi I can help on this. Fran?ois De : freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : lundi 16 janvier 2012 23:54 ? : freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org Objet : [Freeswitch-dev] Nous avons besoin de votre assistance avec les traductions! Salut! If you can read the subject line then you already know what I need: help with some French translations. As you know, our very own Marc Olivier Chouinard (Moc) has produced a set of French-Canadian sound prompts for everyone's benefit. The FreeSWITCH team and Barracuda Networks recently added some prompts to that set. We are looking to expand that sound set even more with some of our newest sound prompts. I have a grand total of about 220 prompts that I would like to get translated into French. Many of these are less-common prompts that a lot of people won't need, but I would like to get the translations done anyway so that we can get the most useful ones recorded as soon as possible. Also, some of these prompts are silly ones that may not translate too well into French. (I've heard that the French don't have a sense of humor but I cannot confirm that report at this time. :) If you'd like to help with this translation project please email me off list and I will send you a spreadsheet with what we have to do. Merci beaucoup pour tout votre soutien! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120117/0701dc7b/attachment.html From msc at freeswitch.org Tue Jan 17 02:28:43 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 15:28:43 -0800 Subject: [Freeswitch-dev] Nous avons besoin de votre assistance avec les traductions! In-Reply-To: <002b01ccd4a5$4d0bc0d0$e7234270$@fr.eu.org> References: <002b01ccd4a5$4d0bc0d0$e7234270$@fr.eu.org> Message-ID: Merci! I've attached a spreadsheet. Anything you can translate would be appreciated. Skip any that you don't feel comfortable trying to translate. Also, I left a column for notes, so if you have a question or comment on the translation please put it there. The last column ("Required?") is for me. Those that are required are the most important ones that I would like to get recorded. Thanks again, Michael On Mon, Jan 16, 2012 at 3:19 PM, wrote: > Hi**** > > ** ** > > I can help on this.**** > > ** ** > > Fran?ois**** > > ** ** > > *De :* freeswitch-dev-bounces at lists.freeswitch.org [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* lundi 16 janvier 2012 23:54 > *? :* freeswitch-users at lists.freeswitch.org; > freeswitch-dev at lists.freeswitch.org > *Objet :* [Freeswitch-dev] Nous avons besoin de votre assistance avec les > traductions!**** > > ** ** > > Salut! > > If you can read the subject line then you already know what I need: help > with some French translations. As you know, our very own Marc Olivier > Chouinard (Moc) has produced a set of French-Canadian sound prompts for > everyone's benefit. The FreeSWITCH team and Barracuda Networks recently > added some prompts to that set. We are looking to expand that sound set > even more with some of our newest sound prompts. > > I have a grand total of about 220 prompts that I would like to get > translated into French. Many of these are less-common prompts that a lot of > people won't need, but I would like to get the translations done anyway so > that we can get the most useful ones recorded as soon as possible. Also, > some of these prompts are silly ones that may not translate too well into > French. (I've heard that the French don't have a sense of humor but I > cannot confirm that report at this time. :) > > If you'd like to help with this translation project please email me off > list and I will send you a spreadsheet with what we have to do. > > Merci beaucoup pour tout votre soutien! > > -MC**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120116/77cc6f57/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: missing-fr-prompts.xls Type: application/vnd.ms-excel Size: 50176 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120116/77cc6f57/attachment-0001.xls From msc at freeswitch.org Tue Jan 17 22:13:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 11:13:21 -0800 Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Tomorrow Message-ID: Hello all! Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18 Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :) Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120117/88e949c7/attachment.html From alexander.lyasin at gmail.com Fri Jan 20 12:28:57 2012 From: alexander.lyasin at gmail.com (Alexander Lyasin) Date: Fri, 20 Jan 2012 14:28:57 +0500 Subject: [Freeswitch-dev] use mod_spandsp Message-ID: <201201201428.57064.alexander.lyasin@gmail.com> How can I use freeswitch for FAX? In 1.0.6 version mod_spandsp module is absent. There are mod_fax and mod_t38gateway, but it not compiled. Thx Alexander. From brian at freeswitch.org Fri Jan 20 19:19:18 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:19:18 -0600 Subject: [Freeswitch-dev] use mod_spandsp In-Reply-To: <201201201428.57064.alexander.lyasin@gmail.com> References: <201201201428.57064.alexander.lyasin@gmail.com> Message-ID: STEP 1. Don't use 1.0.6, (Works great in git head right now, I know personally I tested the hell out of it all) ;) STEP 2. It's Friday... sit back and relax. Join the IRC channel.... /b On Jan 20, 2012, at 3:28 AM, Alexander Lyasin wrote: > How can I use freeswitch for FAX? In 1.0.6 version mod_spandsp module is > absent. There are mod_fax and mod_t38gateway, but it not compiled. > > Thx > > Alexander. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120120/6ca16b34/attachment.html From justlikeef at gmail.com Mon Jan 23 12:57:52 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 23 Jan 2012 04:57:52 -0500 Subject: [Freeswitch-dev] lame and mpg123 Message-ID: <201201230457.53064.justlikeef@gmail.com> The only place I see lame and mpg123 being used is by mod_shout. Is that correct? Thanks, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120123/0489a2fb/attachment.html From anthony.minessale at gmail.com Tue Jan 24 01:43:11 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Jan 2012 16:43:11 -0600 Subject: [Freeswitch-dev] lame and mpg123 In-Reply-To: <201201230457.53064.justlikeef@gmail.com> References: <201201230457.53064.justlikeef@gmail.com> Message-ID: Yes. On Mon, Jan 23, 2012 at 3:57 AM, Rob Hutton wrote: > ** > > The only place I see lame and mpg123 being used is by mod_shout. Is that > correct? > > > Thanks, > > Rob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120123/3afd2119/attachment.html From lconroy at insensate.co.uk Fri Jan 27 00:41:23 2012 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Thu, 26 Jan 2012 21:41:23 +0000 Subject: [Freeswitch-dev] Build issues with today's head on Mac OS X Message-ID: Hi Folks, This may be me, but ... Running this on an Intel MacMini (running Snow Leopard Server 10.6.8): I downloaded a fresh git today, generated the installation & config files, edited modules.conf to add flite & portaudio and fired off make. Installer went off to d/l flite, got the .bz2 file, and then choked; for some reason it had passed the bz file to gunzip rather than bunzip2. Looking at the configure files & getlib.sh the file extension SHOULD have triggered this to be passed to bunzip, but ... On going into libs, zapping he empty flite folder, manually doing bzcat -c -d flite-1.5.1-current.bz2|tar -x and firing up make again, it all worked. version is reported as FreeSWITCH Version 1.0.head (git-7788f3e 2012-01-26 18-01-52 +0000) I'm at a loss: has something changed today? getlib.sh and the generated configure files look like it should do the right thing, but ... So -- With apologies for what may be a FAQ, has anyone any idea of the files to look at to see what's going wrong? [I had two small/fixable extra errors with a TiBook (Power PC, running Leopard), but this one was present with that as well -- that version is shown as FreeSWITCH Version 1.0.head (git-62dc305 2012-01-25 17-56-08 -0600)] all the best, Lawrence From juraj.fabo at gmail.com Mon Jan 30 15:12:38 2012 From: juraj.fabo at gmail.com (Juraj Fabo) Date: Mon, 30 Jan 2012 13:12:38 +0100 Subject: [Freeswitch-dev] ftdm_call_place and caller_data.call_id Message-ID: Hi, Thank you. > > > Hi, >> >> >> You are better off using mod_freetdm.c as the reference, the sample/test > applications may be outdated. > > Unfortunatelly in mod_freetdm.c there seems to be no need for the call_id from the ftdm_call_place_ex(..). > ftdm_channel_t is an opaque type to API users. I don't remember what your > exact question was. You should not use fchan.caller_data, you always get a > copy of caller_data when you need it, the fact that is internally attached > to the fchan pointer is not something you should rely on (and that's why is > in a private header). > > I though that the freetdm generates a new call ID (in context of freetdm) when new call is initated via ftdm_call_place_ex(..) and I was looking for the way how this call_id is announced to the caller application. I've recently tested head from freeswitch git, but the caller_data.call_id is still zero after returning from ftdm_call_place_ex(). My problem is that the comment aroung _ftdm_call_place() in the freetdm api says: * \note When this function returns FTDM_SUCCESS, the member .fchan from caller_data will be set to the channel used to place the call * and .call_id to the generated call id for that call * but my applciation still receives zero. mod_freetdm seems to not care about this call_id at all. So my workaround is stil following: ftdmret = ftdm_call_place_ex(&caller_data, &hunting, usrmsg); if (ftdmret == FTDM_SUCCESS) { callerData = ftdm_channel_get_caller_data(caller_data.fchan); freetdmLibCallId = FtdmCallId(callerData->call_id); } and that was the trigger to my post. Please, is the comment from the freetdm.h about the call_id within the caller_data valid? Is the zero also valid call_id? Thank you in advance Juraj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/3c3a469c/attachment-0001.html From dujinfang at gmail.com Mon Jan 30 15:41:45 2012 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Jan 2012 20:41:45 +0800 Subject: [Freeswitch-dev] msrp support In-Reply-To: References: <5C49D8F0830947CFBA9FC39299FE617C@gmail.com> <1C2F145E-3E1E-4667-8757-03966740CBBB@freeswitch.org> Message-ID: <843C17682CAA4D10A87DF825AA82996C@gmail.com> UPDATE on MSRP As the lib is GPL and incomplete and it crashes when I test with blink, I thought it might be easier to write a simple implementation than debug the existing code. So I have the following code: https://github.com/seven1240/FreeSWITCH/commit/272607bf34d9748f20bd9e8082a1cf95b6f42e2d Please take a look and give some feedback so I can make progress. I followed RFC 4975 with some exceptions: 1) MSRP in FS is always passive, so clients behind NAT can easily connect to it 2) MSRP service only use one port(hardcoded to 8044 for now), so it's easy for firewall configs 3) MSRP always works in B2BUA way, perhaps we can add bypass_media or external MSRP relay later 4) MSRP payload is different than RTP anyway, so I added some apps like msrp_echo, msrp_send, msrp_recv and api like uuid_msrp_send etc. 5) Each session use one TCP connection, no duplex 6) Only support SDP have MSRP only, no mix of audio/videos 7) I tried with blink on Mac and Boghe on Windows, send messages and files with FS is sort of works with the above code. msrp_bridge is buggy but it is possible to send messages but not files for now. So I have new questions for the right way to do this: Current code is coupled in mod_sofia and ugly, but it need stub code into mod_sofia for SDP negotiation anyway, Ideally it should be a new mod_msrp to decouple from mod_sofia, but mod_sofia cannot call functions in mod_msrp I thought about the spandsp fax way but fax using RTP but MSRP using TCP so I don't now if it's possible. Another idea would be put the msrp code into core, the advantage would be we have another media type in core so audio and MSRP data could be mixed in one SDP session. Ideas? Thanks. On Wednesday, December 7, 2011 at 10:11 AM, Seven Du wrote: > http://jira.freeswitch.org/browse/FS-3748 > > any other comments to my other questions? I think it would be a mod_msrp and some msrp_glue.c things. > > > Thanks. > > On Tuesday, December 6, 2011 at 11:49 PM, Brian West wrote: > > Submit this to jira.freeswitch.org (http://jira.freeswitch.org) and we can push things upstream. > > > > /b > > > > On Dec 6, 2011, at 12:08 AM, Seven Du wrote: > > > > > Hi, > > > > > > We are working on msrp support in FS. With the following patch to lib sofia, the msrp SDP can deliver to mod_sofia > > > > > > http://pastebin.freeswitch.org/17944 > > > > > > questions: > > > > > > 1) should I submit this to our jira or to the the sofia team? Any suggestions on the patch? > > > > > > 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, then we need start TCP/MSRP thread to handle media > > > > > > a. It should more like video handling or more like fax ? That means to hack bridge code to also handle msrp, or should we implement api like tx_fax/rx_fax vs tx_msrp/rxmsrp? > > > > > > b. for text over msrp, is it possible to routing the chatplan like mod_sms > > > > > > c. the first stage would be just endpoint to endpoint, shoud we implement relay ? or even switch mode for conference? I'm very new to msrp. > > > > > > > > > 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, I know it's not license compatible, but as I learned from previous talk about video, the code can put elsewhere and we just implement the logic to hook them on. Or other lib we could use? > > > > > > > > > Any suggestion is appreciated. > > > > > > > > > Seven. > > > > > > -- > > > About: http://about.me/dujinfang > > > Blog: http://www.dujinfang.com > > > Proj: http://www.freeswitch.org.cn > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-dev mailing list > > > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > > http://www.freeswitch.org > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/87879eed/attachment.html From msc at freeswitch.org Mon Jan 30 20:09:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 09:09:41 -0800 Subject: [Freeswitch-dev] Community Alert: Our Friend Brian West Is In The Hospital Message-ID: Hello FreeSWITCH Community, We wanted to let everyone know that Brian West is in St. Francis Hospital, Tulsa, OK. He is checked into room 4325. At this point it looks like he will be there for at least a week. I think it's safe to say that Brian is a dynamo not just for FreeSWITCH but for open source VoIP and telephony. My guess is that there's not a person who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't been aided in one way or another by Brian. Personally I can say that in the past five years Brian has helped me immeasurably. I think I have literally asked him thousands of questions and he has patiently answered them all. We would like to have everyone keep Brian in their thoughts and prayers. Please feel free to send him cards and well wishes at the hospital. Also, if you wish to send gifts and such to his home address you may do so: Brian West 714 Osage Ave McAlester, OK 74501-6638 For those who are interested in helping in more practical ways, such as helping to defray the considerable medical costs that no doubt will be incurred, we will have more information shortly. Please spread the word via Facebook, Twitter, and any other medium you have. Let's all let Brian know just how much we value him as a colleague and friend! -Michael Collins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/7e822025/attachment.html From inge.kristiansen at gmail.com Mon Jan 30 12:13:56 2012 From: inge.kristiansen at gmail.com (Inge Kristiansen) Date: Mon, 30 Jan 2012 10:13:56 +0100 Subject: [Freeswitch-dev] Curl Problem Message-ID: Hi everybody I have a small problem from the dial plan. I want to bill calls made by softphones. when I do a curl lookup from the dialplan, I can not find any value in ${curl_response_data}. Testing in the cli returns correct value, so it looks correct. Any suggestions? regds Inge -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/bf8560c0/attachment.html From krice at freeswitch.org Mon Jan 30 20:20:38 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 30 Jan 2012 11:20:38 -0600 Subject: [Freeswitch-dev] Curl Problem In-Reply-To: Message-ID: Inline... Keep in mind that variables in the dialplan are expanded before the first action is even processed... If you want that variable to be usable on the next action w/out a re-entry into the dialplan (ie: transfer) you need to inline that application call... On 1/30/12 3:13 AM, "Inge Kristiansen" wrote: > > Hi everybody > > I have a small problem from the dial plan. I want to bill calls made by > softphones. > when I do a curl lookup from the dialplan, I can not find any value in > ${curl_response_data}. > > Testing in the cli returns correct value, so it looks correct. > > data="data=http://208.117.44.59/LCR/xxxxxxx.aspx?lega=xxxxx&legb=xxxxxx" /> > /> > > Any suggestions? > > regds > Inge > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/a2c71de1/attachment-0001.html From anthony.minessale at gmail.com Mon Jan 30 20:59:31 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 11:59:31 -0600 Subject: [Freeswitch-dev] msrp support In-Reply-To: <843C17682CAA4D10A87DF825AA82996C@gmail.com> References: <5C49D8F0830947CFBA9FC39299FE617C@gmail.com> <1C2F145E-3E1E-4667-8757-03966740CBBB@freeswitch.org> <843C17682CAA4D10A87DF825AA82996C@gmail.com> Message-ID: If its only relevant to sip its ok to live in mod_sofia but it can still be cleanly done. On Mon, Jan 30, 2012 at 6:41 AM, Seven Du wrote: > UPDATE on MSRP > > As the lib is GPL and incomplete and it crashes when I test with blink, I > thought it might be easier to write a simple implementation than debug the > existing code. > > So I have the following code: > > https://github.com/seven1240/FreeSWITCH/commit/272607bf34d9748f20bd9e8082a1cf95b6f42e2d > > Please take a look and give some feedback so I can make progress. > > I followed RFC 4975 with some exceptions: > > 1) MSRP in FS is always passive, so clients behind NAT can easily connect to > it > 2) MSRP service only use one port(hardcoded to 8044 for now), so it's easy > for firewall configs > 3) MSRP always works in B2BUA way, perhaps we can add bypass_media or > external MSRP relay later > 4) MSRP payload is different than RTP anyway, so I added some apps like > msrp_echo, msrp_send, msrp_recv and api like uuid_msrp_send etc. > 5) Each session use one TCP connection, no duplex > 6) Only support SDP have MSRP only, no mix of audio/videos > 7) I tried with blink on Mac and Boghe on Windows, send messages and files > with FS is sort of works with the above code. msrp_bridge is buggy but it is > possible to send messages but not files for now. > > So I have new questions for the right way to do this: > > Current code is coupled in mod_sofia and ugly, but it need stub code into > mod_sofia for SDP negotiation anyway, Ideally it should be a new mod_msrp to > decouple from mod_sofia, but mod_sofia cannot call functions in mod_msrp I > thought about the spandsp fax way but fax using RTP but MSRP using TCP so I > don't now if it's possible. ?Another idea would be put the msrp code into > core, the advantage would be we have another media type in core so audio and > MSRP data could be mixed in one SDP session. > > Ideas? > > Thanks. > > On Wednesday, December 7, 2011 at 10:11 AM, Seven Du wrote: > > http://jira.freeswitch.org/browse/FS-3748 > > any other comments to my other questions? I think it would be ?a mod_msrp > and some msrp_glue.c things. > > > Thanks. > > On Tuesday, December 6, 2011 at 11:49 PM, Brian West wrote: > > Submit this to jira.freeswitch.org and we can push things upstream. > > /b > > On Dec 6, 2011, at 12:08 AM, Seven Du wrote: > > Hi, > > We are working on msrp support in FS. With the following patch to lib sofia, > the msrp SDP can deliver to mod_sofia > > http://pastebin.freeswitch.org/17944 > > questions: > > 1) should I submit this to our jira or to the the sofia team? Any > suggestions on the patch? > > 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, > then we need start TCP/MSRP thread to handle media > > a. It should more like video handling or more like fax ? That means to hack > bridge code to also handle msrp, or should we implement api like > tx_fax/rx_fax vs tx_msrp/rxmsrp? > > b. for text over msrp, is it possible to routing the chatplan like mod_sms > > c. the first stage would be just endpoint to endpoint, shoud we implement > relay ? or even switch mode for conference? I'm very new to msrp. > > > 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, I > know it's not license compatible, but as I learned from previous talk about > video, the code can put elsewhere and we just implement the logic to hook > them on. Or other lib we could use? > > > Any suggestion is appreciated. > > > Seven. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow (http://www.sparrowmailapp.com) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From inge.kristiansen at gmail.com Mon Jan 30 23:32:22 2012 From: inge.kristiansen at gmail.com (Inge Kristiansen) Date: Mon, 30 Jan 2012 21:32:22 +0100 Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 67, Issue 10 In-Reply-To: References: Message-ID: Thank you, Ken. That solved my problem. Awesome. Thank you. 2012/1/30 > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > > Today's Topics: > > 1. Re: msrp support (Seven Du) > 2. Community Alert: Our Friend Brian West Is In The Hospital > (Michael Collins) > 3. Curl Problem (Inge Kristiansen) > 4. Re: Curl Problem (Ken Rice) > > > ---------- Videresendt e-post ---------- > From: Seven Du > To: freeswitch-dev at lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 20:41:45 +0800 > Subject: Re: [Freeswitch-dev] msrp support > UPDATE on MSRP > > As the lib is GPL and incomplete and it crashes when I test with blink, I > thought it might be easier to write a simple implementation than debug the > existing code. > > So I have the following code: > > > https://github.com/seven1240/FreeSWITCH/commit/272607bf34d9748f20bd9e8082a1cf95b6f42e2d > > Please take a look and give some feedback so I can make progress. > > I followed RFC 4975 with some exceptions: > > 1) MSRP in FS is always passive, so clients behind NAT can easily connect > to it > 2) MSRP service only use one port(hardcoded to 8044 for now), so it's easy > for firewall configs > 3) MSRP always works in B2BUA way, perhaps we can add bypass_media or > external MSRP relay later > 4) MSRP payload is different than RTP anyway, so I added some apps like > msrp_echo, msrp_send, msrp_recv and api like uuid_msrp_send etc. > 5) Each session use one TCP connection, no duplex > 6) Only support SDP have MSRP only, no mix of audio/videos > 7) I tried with blink on Mac and Boghe on Windows, send messages and files > with FS is sort of works with the above code. msrp_bridge is buggy but it > is possible to send messages but not files for now. > > So I have new questions for the right way to do this: > > Current code is coupled in mod_sofia and ugly, but it need stub code into > mod_sofia for SDP negotiation anyway, Ideally it should be a new mod_msrp > to decouple from mod_sofia, but mod_sofia cannot call functions in mod_msrp > I thought about the spandsp fax way but fax using RTP but MSRP using TCP so > I don't now if it's possible. Another idea would be put the msrp code into > core, the advantage would be we have another media type in core so audio > and MSRP data could be mixed in one SDP session. > > Ideas? > > Thanks. > > On Wednesday, December 7, 2011 at 10:11 AM, Seven Du wrote: > > http://jira.freeswitch.org/browse/FS-3748 > > any other comments to my other questions? I think it would be a mod_msrp > and some msrp_glue.c things. > > > Thanks. > > On Tuesday, December 6, 2011 at 11:49 PM, Brian West wrote: > > Submit this to jira.freeswitch.org and we can push things upstream. > > /b > > On Dec 6, 2011, at 12:08 AM, Seven Du wrote: > > Hi, > > We are working on msrp support in FS. With the following patch to lib > sofia, the msrp SDP can deliver to mod_sofia > > http://pastebin.freeswitch.org/17944 > > questions: > > 1) should I submit this to our jira or to the the sofia team? Any > suggestions on the patch? > > 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, > then we need start TCP/MSRP thread to handle media > > a. It should more like video handling or more like fax ? That means to > hack bridge code to also handle msrp, or should we implement api like > tx_fax/rx_fax vs tx_msrp/rxmsrp? > > b. for text over msrp, is it possible to routing the chatplan like mod_sms > > c. the first stage would be just endpoint to endpoint, shoud we implement > relay ? or even switch mode for conference? I'm very new to msrp. > > > 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, > I know it's not license compatible, but as I learned from previous talk > about video, the code can put elsewhere and we just implement the logic to > hook them on. Or other lib we could use? > > > Any suggestion is appreciated. > > > Seven. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow (http://www.sparrowmailapp.com) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > ---------- Videresendt e-post ---------- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org, > freeswitch-dev at lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 09:09:41 -0800 > Subject: [Freeswitch-dev] Community Alert: Our Friend Brian West Is In The > Hospital > Hello FreeSWITCH Community, > > We wanted to let everyone know that Brian West is in St. Francis Hospital, > Tulsa, OK. He is checked into room 4325. At this point it looks like he > will be there for at least a week. > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH > but for open source VoIP and telephony. My guess is that there's not a > person who's worked with Asterisk or FreeSWITCH in the past 10 years who > hasn't been aided in one way or another by Brian. Personally I can say that > in the past five years Brian has helped me immeasurably. I think I have > literally asked him thousands of questions and he has patiently answered > them all. > > We would like to have everyone keep Brian in their thoughts and prayers. > Please feel free to send him cards and well wishes at the hospital. Also, > if you wish to send gifts and such to his home address you may do so: > > Brian West > 714 Osage Ave > McAlester, OK > 74501-6638 > > For those who are interested in helping in more practical ways, such as > helping to defray the considerable medical costs that no doubt will be > incurred, we will have more information shortly. > > Please spread the word via Facebook, Twitter, and any other medium you > have. Let's all let Brian know just how much we value him as a colleague > and friend! > > -Michael Collins > > > > ---------- Videresendt e-post ---------- > From: Inge Kristiansen > To: freeswitch-dev at lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 10:13:56 +0100 > Subject: [Freeswitch-dev] Curl Problem > > Hi everybody > > I have a small problem from the dial plan. I want to bill calls made by > softphones. > when I do a curl lookup from the dialplan, I can not find any value in > ${curl_response_data}. > > Testing in the cli returns correct value, so it looks correct. > > > > > Any suggestions? > > regds > Inge > > > > > ---------- Videresendt e-post ---------- > From: Ken Rice > To: "freeswitch-dev at lists.freeswitch.org" < > freeswitch-dev at lists.freeswitch.org> > Cc: > Date: Mon, 30 Jan 2012 11:20:38 -0600 > Subject: Re: [Freeswitch-dev] Curl Problem > Inline... > > Keep in mind that variables in the dialplan are expanded before the first > action is even processed... If you want that variable to be usable on the > next action w/out a re-entry into the dialplan (ie: transfer) you need to > inline that application call... > > > On 1/30/12 3:13 AM, "Inge Kristiansen" wrote: > > > Hi everybody > > I have a small problem from the dial plan. I want to bill calls made by > softphones. > when I do a curl lookup from the dialplan, I can not find any value in > ${curl_response_data}. > > Testing in the cli returns correct value, so it looks correct. > > > > > Any suggestions? > > regds > Inge > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- //Inge Kristiansen onlineSMSC Ltd. Support : (44)2034113453 (47)21953756 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/0563080f/attachment-0001.html From msc at freeswitch.org Mon Jan 30 23:33:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 12:33:02 -0800 Subject: [Freeswitch-dev] UPDATE: Donating To Assist Brian West Message-ID: Good news, We've contacted PayPal and made sure that they are aware of the fact that Brian may be receive a number of donations. Sometimes this can be construed as "suspicious activity" so it's good to be prepared. PayPal has requested that we use payment type "Personal" and that the money is being sent as a "gift." Here's a quick screenshot: Let's all give Brian some PayPal love! Thanks, Michael P.S. - If you have any questions please email me off list. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/d185c999/attachment-0001.html From msc at freeswitch.org Mon Jan 30 23:34:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 12:34:57 -0800 Subject: [Freeswitch-dev] UPDATE: Donating To Assist Brian West In-Reply-To: References: Message-ID: I forgot the most important part! brian at bkw.org Use that email address to send your donations. -MC On Mon, Jan 30, 2012 at 12:33 PM, Michael Collins wrote: > Good news, > > We've contacted PayPal and made sure that they are aware of the fact that > Brian may be receive a number of donations. Sometimes this can be construed > as "suspicious activity" so it's good to be prepared. PayPal has requested > that we use payment type "Personal" and that the money is being sent as a > "gift." Here's a quick screenshot: > > > > Let's all give Brian some PayPal love! > > Thanks, > Michael > > P.S. - If you have any questions please email me off list. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/0361249d/attachment.html From daggelinckxmichel at gmail.com Tue Jan 31 00:04:29 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Mon, 30 Jan 2012 22:04:29 +0100 Subject: [Freeswitch-dev] UPDATE: Donating To Assist Brian West In-Reply-To: References: Message-ID: I dont have a creditcard or paypall account, are there any other options to donate? Michel On Mon, Jan 30, 2012 at 9:34 PM, Michael Collins wrote: > I forgot the most important part! > > brian at bkw.org > > Use that email address to send your donations. > -MC > > > On Mon, Jan 30, 2012 at 12:33 PM, Michael Collins wrote: > >> Good news, >> >> We've contacted PayPal and made sure that they are aware of the fact that >> Brian may be receive a number of donations. Sometimes this can be construed >> as "suspicious activity" so it's good to be prepared. PayPal has requested >> that we use payment type "Personal" and that the money is being sent as a >> "gift." Here's a quick screenshot: >> >> >> >> Let's all give Brian some PayPal love! >> >> Thanks, >> Michael >> >> P.S. - If you have any questions please email me off list. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/6ad0cf04/attachment.html From moises.silva at gmail.com Tue Jan 31 00:17:17 2012 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Jan 2012 16:17:17 -0500 Subject: [Freeswitch-dev] ftdm_call_place and caller_data.call_id In-Reply-To: References: Message-ID: On Mon, Jan 30, 2012 at 7:12 AM, Juraj Fabo wrote: > You are better off using mod_freetdm.c as the reference, the sample/test >> applications may be outdated. >> >> > Unfortunatelly in mod_freetdm.c there seems to be no need for the > call_id from the ftdm_call_place_ex(..). > Yes, FreeSWITCH does not uses that call id for anything. I meant that in general, for anything else, try to use mod_freetdm.c as the reference ... ftdm_channel_t is an opaque type to API users. I don't remember what your >> exact question was. You should not use fchan.caller_data, you always get a >> copy of caller_data when you need it, the fact that is internally attached >> to the fchan pointer is not something you should rely on (and that's why is >> in a private header). >> >> > I though that the freetdm generates a new call ID (in context of freetdm) > when new call is initated via ftdm_call_place_ex(..) and I was looking for > the way how this call_id is announced to the caller application. > I've recently tested head from freeswitch git, but the caller_data.call_id > is still zero after returning from ftdm_call_place_ex(). > That depends on what caller data you are referring to. When returning from ftdm_call_place_ex(), if successful, the caller_data->call_id member should be valid assuming you got the pointer to caller_data from ftdm_channel_get_caller_data(). If you are talking about the caller_data that is provided as an argument to ftdm_call_place_ex() then the call id was not populated there, until now, I just committed a change to populate the call id there too for the user convenience. Additionally, you could wait for the SIGEVENT_DIALING notification, which is the final confirmation that the call is really going out, that notification event contains a valid call id just as any other SIGEVENT_XX event inside the sigmsg structure. You should *always* receive SIGEVENT_DIALING if ftdm_call_place_ex() returned success. In fact, due to the fact we use separate threads, it could be possible that you receive SIGEVENT_DIALING *before* ftdm_call_place_ex() returns, and you should be prepared to handle that case ... > My problem is that the comment aroung _ftdm_call_place() in the freetdm > api says: > > * \note When this function returns FTDM_SUCCESS, the member .fchan from > caller_data will be set to the channel used to place the call > * and .call_id to the generated call id for that call > * > > but my applciation still receives zero. mod_freetdm seems to not care > about this call_id at all. > > This is because you are probably checking the caller_data variable that you provide to ftdm_call_place() not the one returned by ftdm_channel_get_caller_data(). I have improved ftdm_call_place_ex() now to also include the call id in the caller_data argument you provide. > So my workaround is stil following: > > ftdmret = ftdm_call_place_ex(&caller_data, &hunting, usrmsg); > if (ftdmret == FTDM_SUCCESS) > { > callerData = ftdm_channel_get_caller_data(caller_data.fchan); > freetdmLibCallId = FtdmCallId(callerData->call_id); > } > Yes, that's how you were expected to do it before (or wait for SIGEVENT_DIALING) > and that was the trigger to my post. > Please, is the comment from the freetdm.h about the call_id within the > caller_data valid? Is the zero also valid call_id? > The comment is correct, is just talking about the caller data returned by ftdm_channel_caller_data and not the caller data provided as argument to ftdm_call_place_ex(), but with the change I just committed both are valid now. The value 0 is not a valid call_id, call ids start at 1 *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/61b3dc9f/attachment-0001.html From msc at freeswitch.org Tue Jan 31 00:26:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 13:26:30 -0800 Subject: [Freeswitch-dev] UPDATE: Donating To Assist Brian West In-Reply-To: References: Message-ID: Sure. If you can send a check or money order or anything like that then just send it to Brian's home address. You can get all of his contact information at bkw.org. -MC On Mon, Jan 30, 2012 at 1:04 PM, Michel Daggelinckx < daggelinckxmichel at gmail.com> wrote: > I dont have a creditcard or paypall account, are there any other options > to donate? > > Michel > > On Mon, Jan 30, 2012 at 9:34 PM, Michael Collins wrote: > >> I forgot the most important part! >> >> brian at bkw.org >> >> Use that email address to send your donations. >> -MC >> >> >> On Mon, Jan 30, 2012 at 12:33 PM, Michael Collins wrote: >> >>> Good news, >>> >>> We've contacted PayPal and made sure that they are aware of the fact >>> that Brian may be receive a number of donations. Sometimes this can be >>> construed as "suspicious activity" so it's good to be prepared. PayPal has >>> requested that we use payment type "Personal" and that the money is being >>> sent as a "gift." Here's a quick screenshot: >>> >>> >>> >>> Let's all give Brian some PayPal love! >>> >>> Thanks, >>> Michael >>> >>> P.S. - If you have any questions please email me off list. >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/d5703939/attachment.html From dujinfang at gmail.com Tue Jan 31 03:13:56 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Jan 2012 08:13:56 +0800 Subject: [Freeswitch-dev] msrp support In-Reply-To: References: <5C49D8F0830947CFBA9FC39299FE617C@gmail.com> <1C2F145E-3E1E-4667-8757-03966740CBBB@freeswitch.org> <843C17682CAA4D10A87DF825AA82996C@gmail.com> Message-ID: <047CC1E469664CB7B51313EB20452262@gmail.com> Cool. I have moved msrp related applications and apis to msrp.c so it's more clear. Will do more work and update later. Thanks. On Tuesday, January 31, 2012 at 1:59 AM, Anthony Minessale wrote: > If its only relevant to sip its ok to live in mod_sofia but it can > still be cleanly done. > > > On Mon, Jan 30, 2012 at 6:41 AM, Seven Du wrote: > > UPDATE on MSRP > > > > As the lib is GPL and incomplete and it crashes when I test with blink, I > > thought it might be easier to write a simple implementation than debug the > > existing code. > > > > So I have the following code: > > > > https://github.com/seven1240/FreeSWITCH/commit/272607bf34d9748f20bd9e8082a1cf95b6f42e2d > > > > Please take a look and give some feedback so I can make progress. > > > > I followed RFC 4975 with some exceptions: > > > > 1) MSRP in FS is always passive, so clients behind NAT can easily connect to > > it > > 2) MSRP service only use one port(hardcoded to 8044 for now), so it's easy > > for firewall configs > > 3) MSRP always works in B2BUA way, perhaps we can add bypass_media or > > external MSRP relay later > > 4) MSRP payload is different than RTP anyway, so I added some apps like > > msrp_echo, msrp_send, msrp_recv and api like uuid_msrp_send etc. > > 5) Each session use one TCP connection, no duplex > > 6) Only support SDP have MSRP only, no mix of audio/videos > > 7) I tried with blink on Mac and Boghe on Windows, send messages and files > > with FS is sort of works with the above code. msrp_bridge is buggy but it is > > possible to send messages but not files for now. > > > > So I have new questions for the right way to do this: > > > > Current code is coupled in mod_sofia and ugly, but it need stub code into > > mod_sofia for SDP negotiation anyway, Ideally it should be a new mod_msrp to > > decouple from mod_sofia, but mod_sofia cannot call functions in mod_msrp I > > thought about the spandsp fax way but fax using RTP but MSRP using TCP so I > > don't now if it's possible. Another idea would be put the msrp code into > > core, the advantage would be we have another media type in core so audio and > > MSRP data could be mixed in one SDP session. > > > > Ideas? > > > > Thanks. > > > > On Wednesday, December 7, 2011 at 10:11 AM, Seven Du wrote: > > > > http://jira.freeswitch.org/browse/FS-3748 > > > > any other comments to my other questions? I think it would be a mod_msrp > > and some msrp_glue.c things. > > > > > > Thanks. > > > > On Tuesday, December 6, 2011 at 11:49 PM, Brian West wrote: > > > > Submit this to jira.freeswitch.org (http://jira.freeswitch.org) and we can push things upstream. > > > > /b > > > > On Dec 6, 2011, at 12:08 AM, Seven Du wrote: > > > > Hi, > > > > We are working on msrp support in FS. With the following patch to lib sofia, > > the msrp SDP can deliver to mod_sofia > > > > http://pastebin.freeswitch.org/17944 > > > > questions: > > > > 1) should I submit this to our jira or to the the sofia team? Any > > suggestions on the patch? > > > > 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, > > then we need start TCP/MSRP thread to handle media > > > > a. It should more like video handling or more like fax ? That means to hack > > bridge code to also handle msrp, or should we implement api like > > tx_fax/rx_fax vs tx_msrp/rxmsrp? > > > > b. for text over msrp, is it possible to routing the chatplan like mod_sms > > > > c. the first stage would be just endpoint to endpoint, shoud we implement > > relay ? or even switch mode for conference? I'm very new to msrp. > > > > > > 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, I > > know it's not license compatible, but as I learned from previous talk about > > video, the code can put elsewhere and we just implement the logic to hook > > them on. Or other lib we could use? > > > > > > Any suggestion is appreciated. > > > > > > Seven. > > > > -- > > About: http://about.me/dujinfang > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:anthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org) > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120131/9bfc2fb3/attachment-0001.html From msc at freeswitch.org Tue Jan 31 06:01:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 19:01:30 -0800 Subject: [Freeswitch-dev] Update on Brian's Condition Message-ID: Hey all, Brian says that the latest is as follows: The lump in his chest is actually a lymph node. They are running further tests to see if they can track down what's going with it. The mass in his left lung appears just to be an infection, but again they are running more tests to make sure. Please everyone keep Brian in your thoughts and prayers. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120130/48643e59/attachment.html From dujinfang at gmail.com Tue Jan 31 12:51:26 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Jan 2012 17:51:26 +0800 Subject: [Freeswitch-dev] switch_stun_random_string returns identical strings on two calls randomly Message-ID: Hi, I tried to replace the following function which I copied somewhere in msrp.c with switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd615) but I found sometimes switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd615) returns identical strings on two sequential calls, with the following function I never found, so I still use it. By looking the code I understand srand seed is time based so it's possible to return duplicate random string, but I found switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd615) used in some places other than stun, just curious would it cause problems? void random_string(char *buf, switch_size_t size) { long val[4]; int x; for (x = 0; x < 4; x++) val[x] = random(); snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]); *(buf+size) = '\0'; } Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120131/77e67164/attachment.html From anthony.minessale at gmail.com Tue Jan 31 19:03:42 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Jan 2012 10:03:42 -0600 Subject: [Freeswitch-dev] switch_stun_random_string returns identical strings on two calls randomly In-Reply-To: References: Message-ID: ya, you're right. Fixed in tree.... On Tue, Jan 31, 2012 at 3:51 AM, Seven Du wrote: > Hi, > > I tried to replace the following function which I copied somewhere in msrp.c > with ?switch_stun_random_string?but I found > sometimes?switch_stun_random_string?returns identical strings on two > sequential calls, with the following function I never found, so I still use > it. > > By looking the code I understand srand seed is time based so it's possible > to return duplicate random string, but I > found?switch_stun_random_string?used in some places other than stun, just > curious would it cause problems? > > void random_string(char *buf, switch_size_t size) > { > long val[4]; > int x; > > for (x = 0; x < 4; x++) > val[x] = random(); > snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]); > *(buf+size) = '\0'; > } > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: ?http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From vargas at khomp.com.br Tue Jan 31 23:31:40 2012 From: vargas at khomp.com.br (Rafael Vargas) Date: Tue, 31 Jan 2012 18:31:40 -0200 Subject: [Freeswitch-dev] Absolute conditions are not changing the proceed variable... (mod_dialplan_xml.c) Message-ID: Hi, I've noticed that when I have an extension containing only an absolute condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...) is not changed and the next extension is also executed, thus ignoring the first extension's "continue" attribute, which was set to "false". I'm referring to this code: (mod_dialplan.c:221 in today's GIT) ... } else if (time_match == -1) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG, "Dialplan: %s Absolute Condition [%s] match=%s\n", switch_channel_get_name(channel), exten_name, all ? "all" : "any"); pass++; if (!all && !xor) break; } else if (time_match == 1) { ... Shouldn't this set the proceed variable to 1, just like the regex condition a few lines above (line 208 in today's GIT)? I just made a quick test and it worked as (I) expected. What I'm proposing is to change the code above to the following: ... } else if (time_match == -1) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG, "Dialplan: %s Absolute Condition [%s] match=%s\n", switch_channel_get_name(channel), exten_name, all ? "all" : "any"); proceed = 1; pass++; if (!all && !xor) break; } else if (time_match == 1) { ... Sorry if this was already discussed, I've searched this list and the users list to no avail... Best regards, Rafael Scheffer Vargas desenvolvimento KHOMP - 15 Anos R. Joe Colla?o, 253 www.khomp.com.br Florian?polis | SC +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535 vargas at khomp.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120131/09771cb6/attachment.html