From anthony.minessale at gmail.com Mon Jan 4 08:16:08 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 4 Jan 2010 10:16:08 -0600 Subject: [Freeswitch-dev] thesis on live sip calls migration In-Reply-To: <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com> <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> Message-ID: <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> Hi, When would you like to continue this conversation? On Fri, Dec 18, 2009 at 4:38 AM, stefano bossi wrote: > Hi all, > > as promised in #freeswitch-dev I'll try to explain better what I've done > and I would like to achieve for my thesis. > > > At the moment I successfully patched sofsip_cli and now it can do something > near active call migration :) > Thanks to the versatility of nua api I could achieve my objective with > little effort. > Basically I call someone, then I kill sofsip_cli and I restart it. I launch > my "reinvite" command and the call can continue normally. The called client > doesn't realize what happened. > In this case I write call data in a file and when I launch the reinvite I > read directly from this file. > Sofsip_cli sends an INVITE message with some TAG appended to the > nua_handle. So the called phone thinks to receive an in-session invite and > simply refreshes the call, but Sofsip_cli instead allocates all the stuff > for a new call. In this way I can avoid to modify directly the RTP part. It > seems to be all simpler. > > I even wrote a little module for FS able to monitor(using the SIP OPTION > message) the life of a specific sofia profile.. It's just a proof of concept > but it can do failover(without live call migration) between 2 machines in > about 50ms. > This is possible with these actions: > > - registrations are shared with odbc > > > - on the start of the module(present only on the backup machine): > > > 1. I set an arp rule blocking the arp response for the virtual IP (set > on the primary machine) > 2. I set the virtual IP on the backup machine( but no one knows thanks > to the arp rule and so I can bind to vIP) > 3. I prepare and run the sip profile (on the backup)... loading here > the profile permits to save a lot of time during reaction > > > - on the reaction > > > 1. I remove the arp rule > 2. I send a gratuitous arp request > > As I said in about 50ms sip clients can call again. Maybe this time can > decrease using NETLINK socket for the arp table. > > The union of these 2 works will give us a very fast "live profile > migration" :D > > There some points to discuss: > > - switch_core_session_resurrect_channel: I didn't know this function, I > need to understand what it offers, maybe tomorrow ;) > - the propagation of call state variables: my first idea was to use the > multicast events adding some headers to send all the necessary data (but > anthm proposed XML) > - I thought only to sofia aspects.. > > In next days I'll try to understand where to hook in FS to try these ideas. > When I'll have something ready (and without very very big errors:) I'll be > happy to send you the patch. > This is my first work on something real like FS.. I'm opened to any kind of > suggestions!* > > *please contact me for further clarifications > > Thanks > > Stefano Bossi > > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100104/f570110d/attachment.html From david.varnes at gmail.com Tue Jan 5 05:50:07 2010 From: david.varnes at gmail.com (david varnes) Date: Wed, 6 Jan 2010 00:50:07 +1100 Subject: [Freeswitch-dev] putting files onto files.freeswitch.org? Message-ID: <74a861001001050550y646dbe4aqed512dfb7b0e706f@mail.gmail.com> Hi, First post here -- firstly thanks for a fantastic project/product. I have a java esl client implementation in my contrib/dvarnes and have just posted an announcement to users list. 2 questions: * In case there is any interest, and seeing as how many java users would prefer a binary jar to incorporate in a project, is there any process/rules about putting files onto files.freeswitch.org ? I would propose to put a snapshot jar file up for starters and see if there is any feedback. A release could follow. * Is there any way to generate a full set of events from FS to test the inbound ESL client with ? ie do you have some magic incantation that makes FS spit out a set of events ? thanks davidv From tonhudung at gmail.com Wed Jan 6 07:20:16 2010 From: tonhudung at gmail.com (Alex To) Date: Wed, 6 Jan 2010 23:20:16 +0800 Subject: [Freeswitch-dev] Need some explainations about method switch_core_session_run(switch_core_session_t *session) in switch_core_state_machine.c Message-ID: <002201ca8ee3$c3ff9db0$4bfed910$@com> Hi, First of all, pardon my English. English isn?t my native language J. I am reading the method switch_core_session_run(switch_core_session_t *session) and I would be really appreciated if someone could shed some light on how the method works. Below is the logic according to my understanding: 1. LINE 303: while ((state = switch_channel_get_state(session->channel)) != CS_DESTROY) ? if state is CS_DESTROY return and do nothing ? 2. Inside if (state != switch_channel_get_running_state(session->channel) || state >= CS_HANGUP) { } a. LINE 308: state != switch_channel_get_running_state(session->channel) || state >= CS_HANGUP ? to check if channel->state and channel->running_state not equal. This is to avoid running the STATE_MACRO multiple times. However I don?t understand why do we need (state >= CS_HANGUP) ?. b. Inside STATE_MACRO code block i. LINE 229: midstate = state ? to save the current state. ii. LINE 231: if (!driver_state_handler->on_##__STATE || (driver_state_handler->on_##__STATE(session) == SWITCH_STATUS_SUCCESS) ? to check either endpoint_interface->state_handler is NULL or return SWITCH_STATUS_SUCCESS so it will continue to execute handlers in session->channel, runtime.state_handlers and core standard state handlers. It means if the condition is not true, channel->state_handlers, runtime.state_handlers and core standard state handlers will NOT be executed at all ? iii. LINE 246: if (!proceed) global_proceed = 0; ? If something wrong in the previous while loop, do not run core standard state handlers ? iv. LINE 247: proceed = 1; LINE 248: while (do_extra_handlers && proceed && (application_state_handler = switch_core_get_state_handler(index++)) != 0) { } ? Since we set proceed = 1, do we still need (&& proceed) inside the condition in while loop ? v. LINE 260: if (!proceed || midstate != switch_channel_get_state(session->channel)) global_proceed = 0; ? If not proceed or the channel state has changed by some of the state handlers executed earlier, do not execute core standard state handler ? vi. LINE 266: while (silly) ? silly is never changed, does it mean we could go into an endless while loop ? Sorry for the long email and some questions are probably dumb but I am trying to understand about FreeSwitch, especially the core and the core state machine. If it is documented somewhere about the workflow happening inside FreeSwitch, please enlighten me as what I am doing now is to read the code line by line and figure out which may not be the most efficient . Thank you for reading this Regards Alex To -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100106/74b51d00/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 6 08:16:22 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 Jan 2010 10:16:22 -0600 Subject: [Freeswitch-dev] Need some explainations about method switch_core_session_run(switch_core_session_t *session) in switch_core_state_machine.c In-Reply-To: <002201ca8ee3$c3ff9db0$4bfed910$@com> References: <002201ca8ee3$c3ff9db0$4bfed910$@com> Message-ID: <191c3a031001060816t6c27f1f7q7eda1001d87dde9@mail.gmail.com> On Wed, Jan 6, 2010 at 9:20 AM, Alex To wrote: > Hi, > > > > First of all, pardon my English. English isn?t my native language J. > > > > I am reading the method switch_core_session_run(switch_core_session_t > *session) and I would be really appreciated if someone could shed some light > on how the method works. Below is the logic according to my understanding: > > > > 1. LINE 303: while ((state = > switch_channel_get_state(session->channel)) != CS_DESTROY) > > ? if state is CS_DESTROY return and do nothing ? > > > state typically starts at CS_NEW the loop runs until state destroy is reached > 2. Inside if (state != > switch_channel_get_running_state(session->channel) || state >= CS_HANGUP) > > { > > ? > > } > > a. LINE 308: state != > switch_channel_get_running_state(session->channel) || state >= CS_HANGUP > > ? to check if channel->state and channel->running_state not equal. This > is to avoid running the STATE_MACRO multiple times. However I don?t > understand why do we need (state >= CS_HANGUP) ?. > Once we are >= HANGUP we dont want to do that test anymore. > b. Inside STATE_MACRO code block > > i. LINE > 229: midstate = state > > ? to save the current state. > > ii. LINE > 231: if (!driver_state_handler->on_##__STATE || > (driver_state_handler->on_##__STATE(session) == SWITCH_STATUS_SUCCESS) > > ? to check either endpoint_interface->state_handler is NULL or return > SWITCH_STATUS_SUCCESS so it will continue to execute handlers in > session->channel, runtime.state_handlers and core standard state handlers. > It means if the condition is not true, channel->state_handlers, > runtime.state_handlers and core standard state handlers will NOT be executed > at all ? > > iii. LINE > 246: if (!proceed) global_proceed = 0; > > ? If something wrong in the previous while loop, do not run core standard > state handlers ? > yes the endpoints have the power to veto state handlers from being called > iv. LINE > 247: proceed = > 1; > > > LINE 248: while (do_extra_handlers && proceed && > (application_state_handler = switch_core_get_state_handler(index++)) != 0) { > ?} > > ? Since we set proceed = 1, do we still need (&& proceed) inside the > condition in while loop ? > yes because proceed can become 0 within that loop > v. LINE > 260: if (!proceed || midstate != > switch_channel_get_state(session->channel)) global_proceed = 0; > > ? If not proceed or the channel state has changed by some of the state > handlers executed earlier, do not execute core standard state handler ? > Again, veto power. > vi. LINE > 266: while (silly) > > ? silly is never changed, does it mean we could go into an endless while > loop ? > > > Its to stop the compiler from getting mad, that's why it's called silly its specific to using a while loop inside a macro > Sorry for the long email and some questions are probably dumb but I am > trying to understand about FreeSwitch, especially the core and the core > state machine. If it is documented somewhere about the workflow happening > inside FreeSwitch, please enlighten me as what I am doing now is to read the > code line by line and figure out which may not be the most efficient . > > > Why are you digging into this code? do you have some problem you are attempting to solve? The most important thing is that code like this that has taken so long to perfect, even if there could be improvements, is hard to even look at without causing regressions. So I'm curious what you were looking for. > Thank you for reading this > > > > Regards > > > > Alex To > > > > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100106/74405c51/attachment.html From tonhudung at gmail.com Wed Jan 6 09:01:17 2010 From: tonhudung at gmail.com (Alex To) Date: Thu, 7 Jan 2010 01:01:17 +0800 Subject: [Freeswitch-dev] Need some explainations about method switch_core_session_run(switch_core_session_t *session) in switch_core_state_machine.c In-Reply-To: <191c3a031001060816t6c27f1f7q7eda1001d87dde9@mail.gmail.com> References: <002201ca8ee3$c3ff9db0$4bfed910$@com> <191c3a031001060816t6c27f1f7q7eda1001d87dde9@mail.gmail.com> Message-ID: <002d01ca8ef1$e06a3600$a13ea200$@com> Hi, Anthony I appreciate you spend your time answering my questions. I have no problem with FreeSwitch. Since you ask, I am digging into this code because I am trying to understand how FreeSwitch works inside the core, probably will take me a great deal of time but I am trying. I wrote a custom EndPoint in FreeSwitch before to run a modem created by the hardware engineer team. Basically it is similar to Zaptel compatible card. The Modem Endpoint runs in FreeSwitch to provide PSTN Connectivity that uses our specific modem. While working with FreeSwitch, I have this idea to implement something similar to FreeSwitch in 100% .NET managed code. Of course I am not expecting it to be as mature as FreeSwitch but something basic first. A core that allows me to write pluggable modules, basic dial plan and the the first supported endpoint would be XMMP since I had some experience working with this XMMP .Net library before. The ultimate aim is code readability, simplicity and maintainability. I have some other aims too but basically this is it, mostly. I am working on this as a personal hobby project only J. Cheers Alex To From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, January 07, 2010 12:16 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Need some explainations about method switch_core_session_run(switch_core_session_t *session) in switch_core_state_machine.c On Wed, Jan 6, 2010 at 9:20 AM, Alex To wrote: Hi, First of all, pardon my English. English isn?t my native language J. I am reading the method switch_core_session_run(switch_core_session_t *session) and I would be really appreciated if someone could shed some light on how the method works. Below is the logic according to my understanding: 1. LINE 303: while ((state = switch_channel_get_state(session->channel)) != CS_DESTROY) ? if state is CS_DESTROY return and do nothing ? state typically starts at CS_NEW the loop runs until state destroy is reached 2. Inside if (state != switch_channel_get_running_state(session->channel) || state >= CS_HANGUP) { } a. LINE 308: state != switch_channel_get_running_state(session->channel) || state >= CS_HANGUP ? to check if channel->state and channel->running_state not equal. This is to avoid running the STATE_MACRO multiple times. However I don?t understand why do we need (state >= CS_HANGUP) ?. Once we are >= HANGUP we dont want to do that test anymore. b. Inside STATE_MACRO code block i. LINE 229: midstate = state ? to save the current state. ii. LINE 231: if (!driver_state_handler->on_##__STATE || (driver_state_handler->on_##__STATE(session) == SWITCH_STATUS_SUCCESS) ? to check either endpoint_interface->state_handler is NULL or return SWITCH_STATUS_SUCCESS so it will continue to execute handlers in session->channel, runtime.state_handlers and core standard state handlers. It means if the condition is not true, channel->state_handlers, runtime.state_handlers and core standard state handlers will NOT be executed at all ? iii. LINE 246: if (!proceed) global_proceed = 0; ? If something wrong in the previous while loop, do not run core standard state handlers ? yes the endpoints have the power to veto state handlers from being called iv. LINE 247: proceed = 1; LINE 248: while (do_extra_handlers && proceed && (application_state_handler = switch_core_get_state_handler(index++)) != 0) { } ? Since we set proceed = 1, do we still need (&& proceed) inside the condition in while loop ? yes because proceed can become 0 within that loop v. LINE 260: if (!proceed || midstate != switch_channel_get_state(session->channel)) global_proceed = 0; ? If not proceed or the channel state has changed by some of the state handlers executed earlier, do not execute core standard state handler ? Again, veto power. vi. LINE 266: while (silly) ? silly is never changed, does it mean we could go into an endless while loop ? Its to stop the compiler from getting mad, that's why it's called silly its specific to using a while loop inside a macro Sorry for the long email and some questions are probably dumb but I am trying to understand about FreeSwitch, especially the core and the core state machine. If it is documented somewhere about the workflow happening inside FreeSwitch, please enlighten me as what I am doing now is to read the code line by line and figure out which may not be the most efficient . Why are you digging into this code? do you have some problem you are attempting to solve? The most important thing is that code like this that has taken so long to perfect, even if there could be improvements, is hard to even look at without causing regressions. So I'm curious what you were looking for. Thank you for reading this Regards Alex To _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100107/0b9c3fa2/attachment-0001.html From Claudio.Cavalera at italtel.it Thu Jan 7 07:47:33 2010 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 7 Jan 2010 16:47:33 +0100 Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library(off-topic) In-Reply-To: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com> Message-ID: It sounds interesting, but I do not understand how can be useful with freeswitch. Kind Regards, Claudio ________________________________ From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Paulo Pizarro Sent: Thursday, December 24, 2009 3:30 PM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library(off-topic) Hi all, We like to present to all Sofia SIP developers a Lua bind to Sofia SIP, Luasofia. The objective of this project is to be able to use Sofia on a more high level language, Lua was chosen because of its easy integration with C, a lot of our projects are starting to use Lua integrated with C to make development and maintenance easier and faster, because of that we wanted to have a good SIP library to use in Lua, the decision was to make a bind to the already mature Sofia library. The bind is far from being complete, we only bonded the code that we need right now, there's a lot of work to do. We expect that the project will be useful, so a lot of documentation was made to help someone who wants to bind more Sofia code. There is good documentation on how to use it too. To develop we used git to make version control, and we hosted the project at github: http://github.com/ppizarro/luasofia The wiki is at github too: http://wiki.github.com/ppizarro/luasofia We hope Luasofia can be useful to a lot of people, and eventually we get some help making it better and more complete. Best regards, Luasofia team 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/20100107/6ce84134/attachment.html From paulo.pizarro at gmail.com Thu Jan 7 08:52:49 2010 From: paulo.pizarro at gmail.com (Paulo Pizarro) Date: Thu, 7 Jan 2010 14:52:49 -0200 Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library(off-topic) In-Reply-To: References: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com> Message-ID: <8a384d791001070852x164c9a55w1cd4baa43160e792@mail.gmail.com> May be useful for the developers of the module sofia (endpoint SIP). Cheers, Paulo Pizarro 2010/1/7 Cavalera Claudio Luigi > It sounds interesting, > but I do not understand how can be useful with freeswitch. > Kind Regards, > Claudio > > ------------------------------ > *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Paulo Pizarro > *Sent:* Thursday, December 24, 2009 3:30 PM > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip > library(off-topic) > > Hi all, > > We like to present to all Sofia SIP developers a Lua bind to Sofia SIP, > Luasofia. The objective of this project is to be able to use Sofia on a more > high level language, Lua was chosen because of its easy integration with C, > a lot of our projects are starting to use Lua integrated with C to make > development and maintenance easier and faster, because of that we wanted to > have a good SIP library to use in Lua, the decision was to make a bind to > the already mature Sofia library. The bind is far from being complete, we > only bonded the code that we need right now, there's a lot of work to do. We > expect that the project will be useful, so a lot of documentation was made > to help someone who wants to bind more Sofia code. There is good > documentation on how to use it too. > > To develop we used git to make version control, and we hosted the project > at github: > > http://github.com/ppizarro/luasofia > > The wiki is at github too: > > http://wiki.github.com/ppizarro/luasofia > > We hope Luasofia can be useful to a lot of people, and eventually we get > some help making it better and more complete. > > Best regards, > > Luasofia team > > > > 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. > > ******************************************************************************************************************************************** > > _______________________________________________ > 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/20100107/d279f8bf/attachment.html From brian at freeswitch.org Thu Jan 7 08:58:41 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 7 Jan 2010 10:58:41 -0600 Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library(off-topic) In-Reply-To: <8a384d791001070852x164c9a55w1cd4baa43160e792@mail.gmail.com> References: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com> <8a384d791001070852x164c9a55w1cd4baa43160e792@mail.gmail.com> Message-ID: <29C16A62-05F9-4FEC-919C-24E6DAA7A937@freeswitch.org> I can see this possibly replacing SIPP for some more involved test scenarios... maybe write a few? /b On Jan 7, 2010, at 10:52 AM, Paulo Pizarro wrote: > May be useful for the developers of the module sofia (endpoint SIP). > > Cheers, > > Paulo Pizarro From paulo.pizarro at gmail.com Thu Jan 7 10:56:33 2010 From: paulo.pizarro at gmail.com (Paulo Pizarro) Date: Thu, 7 Jan 2010 16:56:33 -0200 Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library(off-topic) In-Reply-To: <29C16A62-05F9-4FEC-919C-24E6DAA7A937@freeswitch.org> References: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com> <8a384d791001070852x164c9a55w1cd4baa43160e792@mail.gmail.com> <29C16A62-05F9-4FEC-919C-24E6DAA7A937@freeswitch.org> Message-ID: <8a384d791001071056r5c9637dbha0d0c678b89d086f@mail.gmail.com> I could do these tests using lua. Is there a specification of which tests to write? 2010/1/7 Brian West > I can see this possibly replacing SIPP for some more involved test > scenarios... maybe write a few? > > /b > > On Jan 7, 2010, at 10:52 AM, Paulo Pizarro wrote: > > > May be useful for the developers of the module sofia (endpoint SIP). > > > > Cheers, > > > > Paulo Pizarro > > > _______________________________________________ > 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/20100107/74aa0c7e/attachment.html From msc at freeswitch.org Thu Jan 7 12:01:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 Jan 2010 12:01:00 -0800 Subject: [Freeswitch-dev] Announcement: FSComm - The FreeSWITCH-based softphone Message-ID: <87f2f3b91001071201vb759713w3f37b192cd09020f@mail.gmail.com> We are happy to announce a new project: FSComm, a FreeSWITCH-based softphone. Read the story here . We look forward to watching this project grow and become a truly useful tool for VoIP users everywhere. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100107/5bc126da/attachment-0001.html From mike at jerris.com Thu Jan 7 19:04:02 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 7 Jan 2010 22:04:02 -0500 Subject: [Freeswitch-dev] putting files onto files.freeswitch.org? In-Reply-To: <74a861001001050550y646dbe4aqed512dfb7b0e706f@mail.gmail.com> References: <74a861001001050550y646dbe4aqed512dfb7b0e706f@mail.gmail.com> Message-ID: If you want to put a jar up, you can catch up with me on irc to discuss. The only issues with files.freeswitch.org is that its backed to a content delivery network so it caches files, if you keep the same file name they can take a long time to update. There is no magic event unit tests like that, but they would be nice to have. Mike On Jan 5, 2010, at 8:50 AM, david varnes wrote: > Hi, > First post here -- firstly thanks for a fantastic project/product. > > I have a java esl client implementation in my contrib/dvarnes > and have just posted an announcement to users list. > > 2 questions: > > * In case there is any interest, and seeing as how many > java users would prefer a binary jar to incorporate in a project, > is there any process/rules about putting files onto > files.freeswitch.org ? > I would propose to put a snapshot jar file up for starters > and see if there is any feedback. A release could follow. > > * Is there any way to generate a full set of events from > FS to test the inbound ESL client with ? ie do you have > some magic incantation that makes FS spit out a set > of events ? From msc at freeswitch.org Fri Jan 8 08:57:19 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 Jan 2010 08:57:19 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly! Message-ID: <87f2f3b91001080857l2f61903cw7f92a850f6c27718@mail.gmail.com> Please call in! We'll mingle for a bit and then get started. Agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_08 -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100108/3418b097/attachment.html From ustcorporation at yahoo.com Fri Jan 8 20:08:51 2010 From: ustcorporation at yahoo.com (Darren C.) Date: Fri, 8 Jan 2010 20:08:51 -0800 (PST) Subject: [Freeswitch-dev] recordFile doesn't seem to honor silence_secs Message-ID: <7669.8919.qm@web33004.mail.mud.yahoo.com> Hello, ? Not able to get FS to honor the silence_secs for recordFile. ? This part of application is very simple.? All we?re doing is asking caller to ?Say your name? and then we record this.? Once there is 3 seconds of silence we want to move on with life so the caller doesn?t have to wait for the full 30 seconds or hit a terminating digit.? Just using the command per the instructions like this: ? session.recordFile("sounds/en/us/callie/audio/commands/1234.wav", mycallback, arg, 30, 500, 3); ? Please advise if there is a known bug, if it is I?ve not been able to find it anywhere.? I see bug reports are supposed to be for 1.0.5 but we?re running on Windows using pre-compiled binaries from:? ? http://files.freeswitch.org/windows_installer/ ? ?holding out for 1.0.5 to come out! ? Our configuration: ? Freeswitch version: 1.0.4 OS: 64-Bit Windows Language: Javascript (mod_spidermonkey) Thanks, teldev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100108/fce7c505/attachment.html From dujinfang at gmail.com Sun Jan 10 09:12:05 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 11 Jan 2010 01:12:05 +0800 Subject: [Freeswitch-dev] Fscomm status and issues and patch Message-ID: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Hi, FsComm is really cool. I played and make some improvements, see rev16229 in trunk(in my contrib dir). And note is stll has bugs when sometimes the sound dev list doesn't sync well with FScomm on some edge cases. Also ideally it can be automatically updated like XLite. Because I cannot join the last weekly conference due to network problems so I don't know the results been discussed. Questions? 1) When will we have jira for this? 2) What the next step of FsComm dev? 3) My time doesn't match well with most of you, so, hard to chat on IRC. Even we have jira for this it would be a pain if I want to commit more patches. Is that ok I commit to my contrib and someone merge to trunk? 4) Is it possible I commit to trunk? Then how to do code review before commit? It's the first time I program QT so some code must be very ugly. Some bugs I found: 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing ">", I guess it's a XML parser bug when it see "/" it expects an ">" to be a "/>". Sorry I deleted the full log but I think it can be replicated so let me know if I need to report to jira. 2) I change event_socket.conf.xml to use port 8022 to avoid conflict with another FS instance. 10% of times I run FsComm it cannot been read. 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template event_socket.conf as follows:
2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template post_load_switch.conf could not be read! 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of event_socket.conf failed 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! Could not listen on 127.0.0.1:8021 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! Could not listen on 127.0.0.1:8021 Cheers. Seven. From jeff at jefflenk.com Sun Jan 10 13:00:40 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 10 Jan 2010 15:00:40 -0600 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: http://jira.freeswitch.org/browse/FSPHONE > Date: Mon, 11 Jan 2010 01:12:05 +0800 > From: dujinfang at gmail.com > To: freeswitch-dev at lists.freeswitch.org > Subject: [Freeswitch-dev] Fscomm status and issues and patch > > Hi, > > FsComm is really cool. I played and make some improvements, see > rev16229 in trunk(in my contrib dir). And note is stll has bugs when > sometimes the sound dev list doesn't sync well with FScomm on some > edge cases. Also ideally it can be automatically updated like XLite. > > Because I cannot join the last weekly conference due to network > problems so I don't know the results been discussed. Questions? > > 1) When will we have jira for this? > 2) What the next step of FsComm dev? > 3) My time doesn't match well with most of you, so, hard to chat on > IRC. Even we have jira for this it would be a pain if I want to commit > more patches. Is that ok I commit to my contrib and someone merge to > trunk? > 4) Is it possible I commit to trunk? Then how to do code review before > commit? It's the first time I program QT so some code must be very > ugly. > > Some bugs I found: > > 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing > ">", I guess it's a XML parser bug when it see "/" it expects an ">" > to be a "/>". Sorry I deleted the full log but I think it can be > replicated so let me know if I need to report to jira. > > 2) I change event_socket.conf.xml to use port 8022 to avoid conflict > with another FS instance. 10% of times I run FsComm it cannot been > read. > > 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template > event_socket.conf as follows: > >
> > > > > > > > >
> > 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template > post_load_switch.conf could not be read! > 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of > event_socket.conf failed > 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > > Cheers. > > Seven. > > _______________________________________________ > 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 _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100110/15cfb13a/attachment.html From jmesquita at freeswitch.org Sun Jan 10 18:40:51 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 10 Jan 2010 23:40:51 -0300 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: Seven, thank you very much for your support. I am sorry for not being very present at the moment but I have been feeling very sick from a virus I contracted. I should be back on my feet tomorrow if anyone cares. Anyway, I will talk to Brian to rename the Jira to FSComm instead of FSPHONE and I would love if you could paste a diff for me in there. I will have it reviewed and commited. I have some changes I have made to the local copy that need to play nice with your modifications. Regards, Jo?o Mesquita On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote: > Hi, > > FsComm is really cool. I played and make some improvements, see > rev16229 in trunk(in my contrib dir). And note is stll has bugs when > sometimes the sound dev list doesn't sync well with FScomm on some > edge cases. Also ideally it can be automatically updated like XLite. > > Because I cannot join the last weekly conference due to network > problems so I don't know the results been discussed. Questions? > > 1) When will we have jira for this? > 2) What the next step of FsComm dev? > 3) My time doesn't match well with most of you, so, hard to chat on > IRC. Even we have jira for this it would be a pain if I want to commit > more patches. Is that ok I commit to my contrib and someone merge to > trunk? > 4) Is it possible I commit to trunk? Then how to do code review before > commit? It's the first time I program QT so some code must be very > ugly. > > Some bugs I found: > > 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing > ">", I guess it's a XML parser bug when it see "/" it expects an ">" > to be a "/>". Sorry I deleted the full log but I think it can be > replicated so let me know if I need to report to jira. > > 2) I change event_socket.conf.xml to use port 8022 to avoid conflict > with another FS instance. 10% of times I run FsComm it cannot been > read. > > 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template > event_socket.conf as follows: > >
> > > > > > > > >
> > 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template > post_load_switch.conf could not be read! > 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of > event_socket.conf failed > 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > > Cheers. > > Seven. > > _______________________________________________ > 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/20100110/ea76fda5/attachment-0001.html From jmesquita at freeswitch.org Sun Jan 10 18:45:11 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 10 Jan 2010 23:45:11 -0300 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: Forgot to reply line by line ... Jo?o Mesquita On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote: > Hi, > > FsComm is really cool. I played and make some improvements, see > rev16229 in trunk(in my contrib dir). And note is stll has bugs when > sometimes the sound dev list doesn't sync well with FScomm on some > edge cases. Also ideally it can be automatically updated like XLite. > > Because I cannot join the last weekly conference due to network > problems so I don't know the results been discussed. Questions? > > 1) When will we have jira for this? > 2) What the next step of FsComm dev? > Have the Sofia accounts configuration pane and right after that, process the events correctly. There are a lot of flaws on how I deal with them at the moment. > 3) My time doesn't match well with most of you, so, hard to chat on > IRC. Even we have jira for this it would be a pain if I want to commit > more patches. Is that ok I commit to my contrib and someone merge to > trunk? > I would love if you could post patches on Jira instead with the work you see fit. Otherwise, I will easily loose track of things. > 4) Is it possible I commit to trunk? Then how to do code review before > commit? It's the first time I program QT so some code must be very > ugly. > > The question above replies this one too. Time will get you to trunk. :-) > Some bugs I found: > > 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing > ">", I guess it's a XML parser bug when it see "/" it expects an ">" > to be a "/>". Sorry I deleted the full log but I think it can be > replicated so let me know if I need to report to jira. > > Please do. This is a bug in switch_event_expand_headers() and I haven't yet had the time to figure out what is the char that breaks it. > 2) I change event_socket.conf.xml to use port 8022 to avoid conflict > with another FS instance. 10% of times I run FsComm it cannot been > read. > The idea is to disable this after some time since it makes no sense at all to be there other then debugging. So I guess you can do whatever with that template. > > 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template > event_socket.conf as follows: > >
> > > > > > > > >
> > 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template > post_load_switch.conf could not be read! > 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of > event_socket.conf failed > 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! > Could not listen on 127.0.0.1:8021 > > Cheers. > > Seven. > > _______________________________________________ > 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/20100110/ecad851a/attachment.html From dujinfang at gmail.com Sun Jan 10 21:48:34 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 11 Jan 2010 13:48:34 +0800 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: <23f91031001102148i543c925euca4517500414882b@mail.gmail.com> Thanks, see inline comments: 2010/1/11 Jo?o Mesquita : > > Forgot to reply line by line ... > > Jo?o Mesquita > > > On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote: >> >> Hi, >> >> FsComm is really cool. I played and make some improvements, see >> rev16229 in trunk(in my contrib dir). And note is stll has bugs when >> sometimes the sound dev list doesn't sync well with FScomm on some >> edge cases. Also ideally it can be automatically updated like XLite. >> >> Because I cannot join the last weekly conference due to network >> problems so I don't know the results been discussed. Questions? >> >> 1) When will we have jira for this? >> 2) What the next step of FsComm dev? > > Have the Sofia accounts configuration pane and right after that, process the > events correctly. There are a lot of flaws on how I deal with them at the > moment. > >> >> 3) My time doesn't match well with most of you, so, hard to chat on >> IRC. Even we have jira for this it would be a pain if I want to commit >> more patches. Is that ok I commit to my contrib and someone merge to >> trunk? > > I would love if you could post patches on Jira instead with the work you see > fit. Otherwise, I will easily loose track of things. > ok, will do. Someone contribute Paypal and I would like to contribute code. Ideally there be a feature list on wiki or roadmap on jira we can follow to avoid invent wheels at the same time, something like this: goal: As Brian described on list: Easy to use, a really nice phone with lots of options, would be kick ass. Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like most softphones do. features: * manually register/unregister for each accounts, very useful for debugging, as in Zoiper * easy dial out through different gateways * auto reconfigure sound devices when dev changes(say a USB headset pluged/unpluged, as in XLite) * Just found a good feature list as I'm writing this list: http://www.zoiper.com/feature_list_zoiper_communicator.php and it would be cool if all "BIZ ONLY" features can be replaced by "FREE" :) Despite the rich features, we should also make it *Extremely* easy to use for non-technical people, E.g. for sip gateway config it only needs username/password/realm, all other params should be in Advanced/Ad-Advanced conf ... >> >> 4) Is it possible I commit to trunk? Then how to do code review before >> commit? It's the first time I program QT so some code must be very >> ugly. >> > > The question above replies this one too. Time will get you to trunk. :-) > >> >> Some bugs I found: >> >> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing >> ">", I guess it's a XML parser bug when it see "/" it expects an ">" >> to be a "/>". Sorry I deleted the full log but I think it can be >> replicated so let me know if I need to report to jira. >> > > Please do. This is a bug in switch_event_expand_headers() and I haven't yet > had the time to figure out what is the char that breaks it. > >> >> 2) I change event_socket.conf.xml to use port 8022 to avoid conflict >> with another FS instance. 10% of times I run FsComm it cannot been >> read. > > The idea is to disable this after some time since it makes no sense at all > to be there other then debugging. So I guess you can do whatever with that > template. > For debugging and for fan, I'd like to keep it. Provide an api allow people make better use of it. Also I noticed API is a "BIZ ONLY" feature on http://www.zoiper.com/feature_list_zoiper_communicator.php >> >> 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template >> event_socket.conf as follows: >> >> ? ?
>> >> ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? >> ? ? >> ?
>> >> 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template >> post_load_switch.conf could not be read! >> 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of >> event_socket.conf failed >> 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! >> Could not listen on 127.0.0.1:8021 >> 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! >> Could not listen on 127.0.0.1:8021 >> >> Cheers. >> >> Seven. >> >> _______________________________________________ >> 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 > > From dujinfang at gmail.com Sun Jan 10 22:02:21 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 11 Jan 2010 14:02:21 +0800 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: <23f91031001102202k46608dc6t799dfb4aa9eb692e@mail.gmail.com> added: http://jira.freeswitch.org/browse/FSPHONE-1 Change to FSComm would make it easier to be found .... :) 2010/1/11 Jo?o Mesquita : > Seven, thank you very much for your support. I am sorry for not being very > present at the moment but I have been feeling very sick from a virus I > contracted. I should be back on my feet tomorrow if anyone cares. > > Anyway, I will talk to Brian to rename the Jira to FSComm instead of FSPHONE > and I would love if you could paste a diff for me in there. I will have it > reviewed and commited. I have some changes I have made to the local copy > that need to play nice with your modifications. > > Regards, > > Jo?o Mesquita > > > On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote: >> >> Hi, >> >> FsComm is really cool. I played and make some improvements, see >> rev16229 in trunk(in my contrib dir). And note is stll has bugs when >> sometimes the sound dev list doesn't sync well with FScomm on some >> edge cases. Also ideally it can be automatically updated like XLite. >> >> Because I cannot join the last weekly conference due to network >> problems so I don't know the results been discussed. Questions? >> >> 1) When will we have jira for this? >> 2) What the next step of FsComm dev? >> 3) My time doesn't match well with most of you, so, hard to chat on >> IRC. Even we have jira for this it would be a pain if I want to commit >> more patches. Is that ok I commit to my contrib and someone merge to >> trunk? >> 4) Is it possible I commit to trunk? Then how to do code review before >> commit? It's the first time I program QT so some code must be very >> ugly. >> >> Some bugs I found: >> >> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing >> ">", I guess it's a XML parser bug when it see "/" it expects an ">" >> to be a "/>". Sorry I deleted the full log but I think it can be >> replicated so let me know if I need to report to jira. >> >> 2) I change event_socket.conf.xml to use port 8022 to avoid conflict >> with another FS instance. 10% of times I run FsComm it cannot been >> read. >> >> 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template >> event_socket.conf as follows: >> >> ? ?
>> >> ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? >> ? ? >> ?
>> >> 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template >> post_load_switch.conf could not be read! >> 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of >> event_socket.conf failed >> 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! >> Could not listen on 127.0.0.1:8021 >> 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! >> Could not listen on 127.0.0.1:8021 >> >> Cheers. >> >> Seven. >> >> _______________________________________________ >> 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 > > From dujinfang at gmail.com Sun Jan 10 22:41:03 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 11 Jan 2010 14:41:03 +0800 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> Message-ID: <23f91031001102241w2c6bec71h6f41d1709a928b39@mail.gmail.com> >> Some bugs I found: >> >> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing >> ">", I guess it's a XML parser bug when it see "/" it expects an ">" >> to be a "/>". Sorry I deleted the full log but I think it can be >> replicated so let me know if I need to report to jira. >> > > Please do. This is a bug in switch_event_expand_headers() and I haven't yet > had the time to figure out what is the char that breaks it. > Haven't do that since I cannot replicate in normal FS but in FSComm. Error msg follows with default settings, home it should be ok when you make all sofia configs done: 2010-01-08 18:20:49.276458 [INFO] mod_qsettings.cpp:51 Template sofia.conf.xml, doesn't exist on directory, copying embedded template. 2010-01-08 18:20:49.277614 [DEBUG] mod_qsettings.cpp:80 Template sofia.conf as follows:
6. When application is running, there is an error in loading the file observed in the console.... 2010-01-12 13:26:21.771159 [CRIT] switch_loadable_module.c:872 Error Loading module D:\FS\freeswitch-1.0.5-latest\freeswitch-1.0.5-20100104-0400\Release\mod \mod_flite.dll **dll open error [126l] ** Could someone please let me know and correct me if I made a mistake in the process. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100112/ddb7a9b1/attachment-0001.html From brian at freeswitch.org Tue Jan 12 08:43:56 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 12 Jan 2010 10:43:56 -0600 Subject: [Freeswitch-dev] Issue with mod_flite In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138064D43A7@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138064D43A7@nickel.mettonigroup.com> Message-ID: <4D6C4F3B-39E3-421C-BCF2-96D4D06BFA24@freeswitch.org> I'm pretty sure modules.conf doesn't control what is compiled on windows. /b On Jan 12, 2010, at 10:35 AM, Suneel Papineni wrote: > 2. Uncommented asr_tts/mod_flite in modules.conf.in file and commented out all other asr_tts modules. > #asr_tts/mod_unimrcp > asr_tts/mod_flite > #asr_tts/mod_pocketsphinx > #asr_tts/mod_cepstral > #asr_tts/mod_tts_commandline -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100112/743a491e/attachment.html From jeff at jefflenk.com Tue Jan 12 09:00:32 2010 From: jeff at jefflenk.com (Jeff Lenk ) Date: Tue, 12 Jan 2010 17:00:32 +0000 Subject: [Freeswitch-dev] Issue with mod_flite Message-ID: You must select the project in the solution and build from there. Mod_flite is not built by default in the solution -----Original Message----- From: Suneel Papineni Sent: 1/12/2010 4:35:23 PM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue with mod_flite Hi, I am facing a problem with mod_flite. Could someone help me in resolving this issue. 1. I am building a Windows version of Freeswitch and using Freeswitch1.0.5_latest. 2. Uncommented asr_tts/mod_flite in modules.conf.in file and commented out all other asr_tts modules. #asr_tts/mod_unimrcp asr_tts/mod_flite #asr_tts/mod_pocketsphinx #asr_tts/mod_cepstral #asr_tts/mod_tts_commandline 3. Now tried to build a "Debug" or "Release" solution. 4. In both cases after successful build, I am not finding mod_flite module in "mod" folder. 5. Before running Freeswitch.exe, edited "modules.conf.xml" file as follows: 6. When application is running, there is an error in loading the file observed in the console.... 2010-01-12 13:26:21.771159 [CRIT] switch_loadable_module.c:872 Error Loading module D:\FS\freeswitch-1.0.5-latest\freeswitch-1.0.5-20100104-0400\Release\mod\mod_flite.dll **dll open error [126l] ** Could someone please let me know and correct me if I made a mistake in the process. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100112/c3d17e93/attachment.html From Suneel.Papineni at mettoni.com Tue Jan 12 09:27:57 2010 From: Suneel.Papineni at mettoni.com (Suneel Papineni) Date: Tue, 12 Jan 2010 17:27:57 -0000 Subject: [Freeswitch-dev] Issue with mod_flite In-Reply-To: References: Message-ID: <3181A30B8C35AB4AA8577B78DDF46138064D43E2@nickel.mettonigroup.com> Hi Jeff, Thanks for the solution. It works and respective dll got created in "mod" folder. Thanks & Regards Suneel From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: 12 January 2010 17:01 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Issue with mod_flite You must select the project in the solution and build from there. Mod_flite is not built by default in the solution -----Original Message----- From: Suneel Papineni Sent: 1/12/2010 4:35:23 PM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue with mod_flite Hi, I am facing a problem with mod_flite. Could someone help me in resolving this issue. 1. I am building a Windows version of Freeswitch and using Freeswitch1.0.5_latest. 2. Uncommented asr_tts/mod_flite in modules.conf.in file and commented out all other asr_tts modules. #asr_tts/mod_unimrcp asr_tts/mod_flite #asr_tts/mod_pocketsphinx #asr_tts/mod_cepstral #asr_tts/mod_tts_commandline 3. Now tried to build a "Debug" or "Release" solution. 4. In both cases after successful build, I am not finding mod_flite module in "mod" folder. 5. Before running Freeswitch.exe, edited "modules.conf.xml" file as follows: 6. When application is running, there is an error in loading the file observed in the console.... 2010-01-12 13:26:21.771159 [CRIT] switch_loadable_module.c:872 Error Loading module D:\FS\freeswitch-1.0.5-latest\freeswitch-1.0.5-20100104-0400\Release\mod \mod_flite.dll **dll open error [126l] ** Could someone please let me know and correct me if I made a mistake in the process. Thanks & Regards Suneel ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100112/0d874d7e/attachment.html From jmesquita at freeswitch.org Tue Jan 12 12:14:14 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 12 Jan 2010 18:14:14 -0200 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: <23f91031001102148i543c925euca4517500414882b@mail.gmail.com> References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> <23f91031001102148i543c925euca4517500414882b@mail.gmail.com> Message-ID: Seven, thank you for the code! I am just now starting to feel better and I will get back to work. You ideas are exactly what I have been putting in practice lately. Look at the accounts tab. Just add a widget where users can add or remove params. They gotta know the param names to be able to do that. Jo?o Mesquita On Mon, Jan 11, 2010 at 3:48 AM, Seven Du wrote: > Thanks, see inline comments: > > 2010/1/11 Jo?o Mesquita : > > > > Forgot to reply line by line ... > > > > Jo?o Mesquita > > > > > > On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote: > >> > >> Hi, > >> > >> FsComm is really cool. I played and make some improvements, see > >> rev16229 in trunk(in my contrib dir). And note is stll has bugs when > >> sometimes the sound dev list doesn't sync well with FScomm on some > >> edge cases. Also ideally it can be automatically updated like XLite. > >> > >> Because I cannot join the last weekly conference due to network > >> problems so I don't know the results been discussed. Questions? > >> > >> 1) When will we have jira for this? > >> 2) What the next step of FsComm dev? > > > > Have the Sofia accounts configuration pane and right after that, process > the > > events correctly. There are a lot of flaws on how I deal with them at the > > moment. > > > >> > >> 3) My time doesn't match well with most of you, so, hard to chat on > >> IRC. Even we have jira for this it would be a pain if I want to commit > >> more patches. Is that ok I commit to my contrib and someone merge to > >> trunk? > > > > I would love if you could post patches on Jira instead with the work you > see > > fit. Otherwise, I will easily loose track of things. > > > > ok, will do. Someone contribute Paypal and I would like to contribute > code. Ideally there be a feature list on wiki or roadmap on jira we > can follow to avoid invent wheels at the same time, something like > this: > > goal: > > As Brian described on list: > > Easy to use, a really nice phone with lots of options, would be kick ass. > Must work on 32bit and 64bit of Windows, Mac and Linux ... and not > suck like most softphones do. > > features: > > * manually register/unregister for each accounts, very useful for > debugging, as in Zoiper > * easy dial out through different gateways > * auto reconfigure sound devices when dev changes(say a USB headset > pluged/unpluged, as in XLite) > * Just found a good feature list as I'm writing this list: > http://www.zoiper.com/feature_list_zoiper_communicator.php and it > would be cool if all "BIZ ONLY" features can be replaced by "FREE" :) > > Despite the rich features, we should also make it *Extremely* easy to > use for non-technical people, E.g. for sip gateway config it only > needs username/password/realm, all other params should be in > Advanced/Ad-Advanced conf ... > > > >> > >> 4) Is it possible I commit to trunk? Then how to do code review before > >> commit? It's the first time I program QT so some code must be very > >> ugly. > >> > > > > The question above replies this one too. Time will get you to trunk. :-) > > > >> > >> Some bugs I found: > >> > >> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing > >> ">", I guess it's a XML parser bug when it see "/" it expects an ">" > >> to be a "/>". Sorry I deleted the full log but I think it can be > >> replicated so let me know if I need to report to jira. > >> > > > > Please do. This is a bug in switch_event_expand_headers() and I haven't > yet > > had the time to figure out what is the char that breaks it. > > > >> > >> 2) I change event_socket.conf.xml to use port 8022 to avoid conflict > >> with another FS instance. 10% of times I run FsComm it cannot been > >> read. > > > > The idea is to disable this after some time since it makes no sense at > all > > to be there other then debugging. So I guess you can do whatever with > that > > template. > > > > For debugging and for fan, I'd like to keep it. Provide an api allow > people make better use of it. Also I noticed API is a "BIZ ONLY" > feature on http://www.zoiper.com/feature_list_zoiper_communicator.php > > >> > >> 2010-01-10 23:35:53.631687 [DEBUG] mod_qsettings.cpp:80 Template > >> event_socket.conf as follows: > >> > >>
> >> > >> > >> > >> > >> > >> > >> > >> > >>
> >> > >> 2010-01-10 23:35:53.631490 [ERR] mod_qsettings.cpp:62 Template > >> post_load_switch.conf could not be read! > >> 2010-01-10 23:35:53.631799 [ERR] mod_event_socket.c:2543 Open of > >> event_socket.conf failed > >> 2010-01-10 23:35:53.631918 [ERR] mod_event_socket.c:2639 Socket Error! > >> Could not listen on 127.0.0.1:8021 > >> 2010-01-10 23:35:53.732089 [ERR] mod_event_socket.c:2639 Socket Error! > >> Could not listen on 127.0.0.1:8021 > >> > >> Cheers. > >> > >> Seven. > >> > >> _______________________________________________ > >> 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 > > > > > > _______________________________________________ > 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/20100112/74616d65/attachment-0001.html From jmesquita at freeswitch.org Tue Jan 12 12:15:38 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 12 Jan 2010 18:15:38 -0200 Subject: [Freeswitch-dev] Fscomm status and issues and patch In-Reply-To: <23f91031001102241w2c6bec71h6f41d1709a928b39@mail.gmail.com> References: <23f91031001100912w5187908eq4745791b40299548@mail.gmail.com> <23f91031001102241w2c6bec71h6f41d1709a928b39@mail.gmail.com> Message-ID: Like I said, the problem is in switch_event_expand_headers(). I think that the main problem is in the "/" character, but I haven't yet drilled down to that. Regards, Jo?o Mesquita On Mon, Jan 11, 2010 at 4:41 AM, Seven Du wrote: > >> Some bugs I found: > >> > >> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing > >> ">", I guess it's a XML parser bug when it see "/" it expects an ">" > >> to be a "/>". Sorry I deleted the full log but I think it can be > >> replicated so let me know if I need to report to jira. > >> > > > > Please do. This is a bug in switch_event_expand_headers() and I haven't > yet > > had the time to figure out what is the char that breaks it. > > > > Haven't do that since I cannot replicate in normal FS but in FSComm. > Error msg follows with default settings, home it should be ok when you > make all sofia configs done: > > 2010-01-08 18:20:49.276458 [INFO] mod_qsettings.cpp:51 Template > sofia.conf.xml, doesn't exist on directory, copying embedded template. > 2010-01-08 18:20:49.277614 [DEBUG] mod_qsettings.cpp:80 Template > sofia.conf as follows: > >
> > > > > > name="rewrite-multicasted-fs-path" value=""/> > > > > > name="user-agent-string" value="2010-01-08 18:20:49.277666 [ERR] > switch_xml.c:1570 Error[[error near line 13]: m > issing >] > 2010-01-08 18:20:49.277733 [DEBUG] sofia.c:2434 user-agent-string > [FreeSWITCH/FSComm] > > _______________________________________________ > 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/20100112/c279b773/attachment.html From jmesquita at freeswitch.org Wed Jan 13 08:00:25 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 13 Jan 2010 14:00:25 -0200 Subject: [Freeswitch-dev] FSComm settings change Message-ID: I am writing to you to let you know that I have made changes on the mod_qsettings. I gave it another thought while programming the persistency of the accounts and using templates for the configuration is overkill and at this stage of the development, will only cause delays since it requires a lot more coding. I am finishing the coding for the new mod_qsettings tonight and I will commit the latest code. This will automatically solve the only bug posted on Jira nowadays. Please not that, even tho there are changes to mod_qsettings, it will not, by any means, impose limitations to new extra parameters on the configuration files and it will still play nicely with static configuration files for modules that do not have its own configuration pane and therefore are not provided by mod_qsettings. Regards, Jo?o Mesquita FSComm Lead Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100113/814e1359/attachment.html From brian at freeswitch.org Thu Jan 14 09:00:09 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 14 Jan 2010 11:00:09 -0600 Subject: [Freeswitch-dev] mod_java is moving to unsupported on Jan 20th. Message-ID: If nobody steps up to take over maintenance of mod_java by the 20th of January we'll be moving it to unsupported. If you wish to take over maintenance please contact me off list ASAP. Thanks, Brian West FreeSWITCH From msc at freeswitch.org Thu Jan 14 11:51:37 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 Jan 2010 11:51:37 -0800 Subject: [Freeswitch-dev] Announcement: FreeSWITCH Event Socket Library Supports TCL Message-ID: <87f2f3b91001141151w361de47fgc6e1db6f9929ea06@mail.gmail.com> Just to let everyone know, we've added TCL support for ESL. More info here: http://www.freeswitch.org/node/228 Let us know if you do anything interesting with it and we'll add it to the wiki. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100114/f64744fc/attachment.html From msc at freeswitch.org Fri Jan 15 08:57:05 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Jan 2010 08:57:05 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly! Message-ID: <87f2f3b91001150857r5d1ad490ga8d29c3f8bf856b9@mail.gmail.com> The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_05 Come join us and let's talk about FreeSWITCH, VoIP, and all things telephonic! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100115/7194b1f4/attachment.html From plite2012 at gmail.com Fri Jan 15 09:04:34 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 15 Jan 2010 11:04:34 -0600 Subject: [Freeswitch-dev] switch_ivr_originate.c(2621) uninitialized local variable 'b_frame' Message-ID: I am having trouble to build FreeSwitch due to the following error: c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): warning C4701: potentially uninitialized local variable 'b_frame' used c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): error C2220: warning treated as error - no 'object' file generated I am using Visual Studio 2008. I could disable "warning treated as error" to continue compiling. But, I don't know this warning would ultimately cause any trouble later. I am raising this problem for other developers' attention on "potentially uninitialized variable". From mrene_lists at avgs.ca Fri Jan 15 09:36:50 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 15 Jan 2010 12:36:50 -0500 Subject: [Freeswitch-dev] switch_ivr_originate.c(2621) uninitialized local variable 'b_frame' In-Reply-To: References: Message-ID: Try revision 16322. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 15-Jan-10, at 12:04 PM, Paul Li wrote: > I am having trouble to build FreeSwitch due to the following error: > > c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): warning > C4701: potentially uninitialized local variable 'b_frame' used > > c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): error C2220: > warning treated as error - no 'object' file generated > > I am using Visual Studio 2008. I could disable "warning treated as > error" to continue compiling. But, I don't know this warning would > ultimately cause any trouble later. I am raising this problem for > other developers' attention on "potentially uninitialized variable". > > _______________________________________________ > 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 brian at freeswitch.org Fri Jan 15 09:42:55 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 15 Jan 2010 11:42:55 -0600 Subject: [Freeswitch-dev] switch_ivr_originate.c(2621) uninitialized local variable 'b_frame' In-Reply-To: References: Message-ID: <5F918567-3576-47FB-92A1-0ADE9E0A1F90@freeswitch.org> Moving forward issues like this go in jira please. /b On Jan 15, 2010, at 11:36 AM, Mathieu Rene wrote: > Try revision 16322. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 15-Jan-10, at 12:04 PM, Paul Li wrote: > >> I am having trouble to build FreeSwitch due to the following error: >> >> c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): warning >> C4701: potentially uninitialized local variable 'b_frame' used >> >> c:\dev\voip\freeswitch\src\switch_ivr_originate.c(2621): error C2220: >> warning treated as error - no 'object' file generated >> >> I am using Visual Studio 2008. I could disable "warning treated as >> error" to continue compiling. But, I don't know this warning would >> ultimately cause any trouble later. I am raising this problem for >> other developers' attention on "potentially uninitialized variable". >> >> _______________________________________________ >> 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/20100115/400fe4fe/attachment-0001.html From plite2012 at gmail.com Fri Jan 15 16:26:26 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 15 Jan 2010 18:26:26 -0600 Subject: [Freeswitch-dev] FSComm VS2008 Project file - updated Message-ID: The present FSComm VS2008 Project file need to be updated as attached in order to build it with Visual Studio 2008 for the issues: (1) prefaccounts.h/prefaccounts.cpp need to be added to the project. (2) "Generated Files" need to be filtered properly for Debug/Release configuration, so that the proper files are used for the corresponding configuration. (3) Link to proper "FreeSwitchCore.lib", based on the selected configuration. (4) All intermediate files now go to $(IntDir), except for "ui_*.h". All these issues have been corrected with the attached .proj file. Thanks for sharing the app! -------------- next part -------------- A non-text attachment was scrubbed... Name: FSComm.2008_My.vcproj Type: application/octet-stream Size: 30669 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100115/4d1cefcc/attachment-0001.obj From jmesquita at freeswitch.org Fri Jan 15 19:44:22 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 16 Jan 2010 01:44:22 -0200 Subject: [Freeswitch-dev] FSComm VS2008 Project file - updated In-Reply-To: References: Message-ID: Carlos, Jeff, Would any of you take a look at that for us, please? Paul, thank you for the contribution. Would you please, next time, post a Jira so we are able to keep track of all contributions? Regards, Jo?o Mesquita FSComm Lead Developer On Fri, Jan 15, 2010 at 10:26 PM, Paul Li wrote: > The present FSComm VS2008 Project file need to be updated as attached > in order to build it with Visual Studio 2008 for the issues: > > (1) prefaccounts.h/prefaccounts.cpp need to be added to the project. > (2) "Generated Files" need to be filtered properly for Debug/Release > configuration, so that the proper files are used for the corresponding > configuration. > (3) Link to proper "FreeSwitchCore.lib", based on the selected > configuration. > (4) All intermediate files now go to $(IntDir), except for "ui_*.h". > > All these issues have been corrected with the attached .proj file. > > Thanks for sharing the app! > > _______________________________________________ > 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/20100116/5182c5b0/attachment.html From plite2012 at gmail.com Sat Jan 16 23:35:17 2010 From: plite2012 at gmail.com (Paul Li) Date: Sun, 17 Jan 2010 01:35:17 -0600 Subject: [Freeswitch-dev] How to use FSComm to make a call? Message-ID: I have built FSComm on Windows with Visual Studio 2008. I am able to register my VoIP service provider, but I am receiving an error "NO_ROUTE_DESTINATION" when trying to call my home number with "0019876543210". I have tried FreeSwitch with X-Lite before and was able to make the same call. I am wondering where I need to modify my configuration so that I can make a similar call. Any screen print or example of calling is welcome. From jmesquita at freeswitch.org Sun Jan 17 15:14:38 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 17 Jan 2010 21:14:38 -0200 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Paul, I found the problem. It can be divided into 2 parts. 1. I was not setting the default account to be used This is fixed now on rev: 16346 but new GUI changes will be made so we are able to select the default account to be used and also display what account is the default one. 2. mod_portaudio had a bug of setting the default dialplan module to the string "default" which is plain wrong, and we were not overwriting it on FSComm's settings module, causing it to simply, silently, not route the call. This has been fixed on rev: 16345 To get these changes working, you will need to recompile mod_portaudio and FSComm. Thank you for your interest in the project and sorry for the long time to fix that, afterall, it is weekend. Regards, Jo?o Mesquita FSComm Developer On Sun, Jan 17, 2010 at 5:35 AM, Paul Li wrote: > I have built FSComm on Windows with Visual Studio 2008. I am able to > register my VoIP service provider, but I am receiving an error > "NO_ROUTE_DESTINATION" when trying to call my home number with > "0019876543210". I have tried FreeSwitch with X-Lite before and was > able to make the same call. I am wondering where I need to modify my > configuration so that I can make a similar call. > > Any screen print or example of calling is welcome. > > _______________________________________________ > 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/20100117/bd22cc56/attachment.html From plite2012 at gmail.com Sun Jan 17 18:34:33 2010 From: plite2012 at gmail.com (Paul Li) Date: Sun, 17 Jan 2010 20:34:33 -0600 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Jo?o, thank you very much. It is now working like a charm. One small typo is about a normal clearing, "Call with (5000) failed with reason NORMAL_CLEARING", which I believe should be "Call with (5000) ended with ...", as this is not an error. I have one more question on how to tie a remote endpoint to the default IVR at Ext 5000. I used to use the originate command, such as originate sofia/default/1008%192.168.4.151 5000 so that the remote endpoint will be bridged to the announcement at Ext 5000. How can I do this in FSComm? From jmesquita at freeswitch.org Sun Jan 17 18:51:25 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 18 Jan 2010 00:51:25 -0200 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Paul, I am not sure I understood your question. Can you elaborate a little bit better? Also, the code you have right now is buggy and I am working at it as we speak. Right now, settings are not correctly persisted for some reason. I was working on a patch posted on Jira because I didn't want code to move a lot further and make it harder to merge later on. Please be patient since this is a very new project. Also, if you want a more live conversation, you can join #fscomm @ freenode. I am always there. Regards, Jo?o Mesquita FSComm Developer On Mon, Jan 18, 2010 at 12:34 AM, Paul Li wrote: > Jo?o, thank you very much. It is now working like a charm. One small > typo is about a normal clearing, "Call with (5000) failed with reason > NORMAL_CLEARING", which I believe should be "Call with (5000) ended > with ...", as this is not an error. > > I have one more question on how to tie a remote endpoint to the > default IVR at Ext 5000. I used to use the originate command, such as > > originate sofia/default/1008%192.168.4.151 5000 > > so that the remote endpoint will be bridged to the announcement at Ext > 5000. How can I do this in FSComm? > > _______________________________________________ > 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/20100118/aeed1845/attachment.html From jmesquita at freeswitch.org Sun Jan 17 20:23:56 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 18 Jan 2010 02:23:56 -0200 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Paul, Just had the commit that really makes things work (16350 ). If not, please report. Recompile FSComm and you will get new features on it as well contributed by Seven. Regards, Jo?o Mesquita FSComm Developer 2010/1/18 Jo?o Mesquita > Paul, I am not sure I understood your question. Can you elaborate a little > bit better? Also, the code you have right now is buggy and I am working at > it as we speak. > > Right now, settings are not correctly persisted for some reason. I was > working on a patch posted on Jira because I didn't want code to move a lot > further and make it harder to merge later on. > > Please be patient since this is a very new project. > > Also, if you want a more live conversation, you can join #fscomm @ > freenode. I am always there. > > Regards, > > Jo?o Mesquita > FSComm Developer > > > > On Mon, Jan 18, 2010 at 12:34 AM, Paul Li wrote: > >> Jo?o, thank you very much. It is now working like a charm. One small >> typo is about a normal clearing, "Call with (5000) failed with reason >> NORMAL_CLEARING", which I believe should be "Call with (5000) ended >> with ...", as this is not an error. >> >> I have one more question on how to tie a remote endpoint to the >> default IVR at Ext 5000. I used to use the originate command, such as >> >> originate sofia/default/1008%192.168.4.151 5000 >> >> so that the remote endpoint will be bridged to the announcement at Ext >> 5000. How can I do this in FSComm? >> >> _______________________________________________ >> 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/20100118/b419abd3/attachment.html From mike at jerris.com Mon Jan 18 01:15:51 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 18 Jan 2010 04:15:51 -0500 Subject: [Freeswitch-dev] IMPORTANT -- Call for bug updates Message-ID: To all FreeSWITCH developers- If you have bugs open on http://jira.freeswitch.org please login today and post a status update on these bugs (even if they appear to be awaiting comment by the development team). If you have a patch, please update these patches to svn trunk so that they may be reviewed. I know many patches have been sitting for quite some time but I will make a strong effort to review and merge patches that are ready to go in. If you have a bug, please update to svn trunk and comment if the issue still exists or is now fixed in trunk. If you bug has a comment requesting more information, please provide it. If you don't have any open bugs, and are not currently using recent svn trunk, I would appreciate it if you could carve a little bit of time out of your days and test out trunk. Feel free to look through jira and find a bug you are interested in and test it as well. We are working towards having the most stable and feature rich release of FreeSWITCH yet and we need your support and assistance to do so. As always, if you have a new bug to file, please do so as soon as possible and try to get as much information as possible on the bug. There are some good guidelines for reporting at http://wiki.freeswitch.org/wiki/Reporting_Bugs As always, many thanks to the community for all your hard work and support. Mike From plite2012 at gmail.com Mon Jan 18 12:16:15 2010 From: plite2012 at gmail.com (Paul Li) Date: Mon, 18 Jan 2010 14:16:15 -0600 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Jo?o, thank you again. It is working better. Do not bother my previous question. I have solved the issue regarding originate command call with no problem. I just need to adapt my thinking with FreeSwitch embedded vs standalone. I did find a bug related to SIP accounts. I have two SIP service providers, and one need a a custom parameter. However, the custom parameter also show up in the second SIP account's advanced tab (for custom parameter). I checked the registry and find the second provider does not have the custom parameter. So, the bug may be just a GUI thing, which you might forget to clear up. Is the first provider assumed to be the default gateway? From jmesquita at freeswitch.org Mon Jan 18 12:45:48 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 18 Jan 2010 18:45:48 -0200 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: Inline... Jo?o Mesquita FSComm Developer On Mon, Jan 18, 2010 at 6:16 PM, Paul Li wrote: > Jo?o, thank you again. It is working better. Do not bother my previous > question. I have solved the issue regarding originate command call > with no problem. I just need to adapt my thinking with FreeSwitch > embedded vs standalone. > > I did find a bug related to SIP accounts. I have two SIP service > providers, and one need a a custom parameter. However, the custom > parameter also show up in the second SIP account's advanced tab (for > custom parameter). I checked the registry and find the second provider > does not have the custom parameter. So, the bug may be just a GUI > thing, which you might forget to clear up. > Yeah, I will fix that bug tonight. > > Is the first provider assumed to be the default gateway? > Right now, yes, but I am going to let you select tonight using the right click mouse button on the account list on the bottom left corner of the main window. > > _______________________________________________ > 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/20100118/782f7ec3/attachment.html From msc at freeswitch.org Mon Jan 18 13:18:50 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Jan 2010 13:18:50 -0800 Subject: [Freeswitch-dev] Call for help - content ideas for freeswitch.org Message-ID: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com> Hello all, I'd like to ask everyone to think about things that we can put up on freeswitch.org. We are interested in anything related to FreeSWITCH, certainly, but also any VoIP or telecom news/articles on other sites that are of interest to FreeSWITCH users. (See http://www.freeswitch.org/node/229as an example.) Please email me off list if you have ideas about content that would be appropriate for our main page. Likewise, if you would like to be the author of stories and/or blog posts on the main page then definitely let me know. We would love to have some fresh perspectives represented. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100118/18ad41b3/attachment.html From devel at thom.fr.eu.org Mon Jan 18 14:01:19 2010 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Mon, 18 Jan 2010 23:01:19 +0100 Subject: [Freeswitch-dev] openzap information + mod_say_fr contribution Message-ID: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> Hello, First let me congratulate for this great software. I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. I could see the intermitent ringing comes from this line. zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) The second point about CLIP is more difficult. First, let me indicate that I?m residing in France and therefore using carrier and phones complying to european standard. About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. I don?t know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel Seizure is present only in FSK-V23 Protocol. It consists in a series of 90 to 300 alternate SPACE/MARK bits ? Considering this, I guess that I should use the else branch in the test ? zchan->token_count > 1 ?. What does this token_count refer to and where should I set it. I?m still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? Thanks in advance Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100118/0802e280/attachment.html From mike at jerris.com Mon Jan 18 14:18:24 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 18 Jan 2010 17:18:24 -0500 Subject: [Freeswitch-dev] openzap information + mod_say_fr contribution In-Reply-To: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> References: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> Message-ID: In regards to the CLIP. The bulk of this code for V23 as you can see is already there, we just need this troubleshooting as we have never tested it. If you can get a raw recording of the channel it will probably help in testing. As for the mod_say_fr, please file this on http://jira.freeswitch.org Mike On Jan 18, 2010, at 5:01 PM, wrote: > Hello, > > First let me congratulate for this great software. > > I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. > > I could see the intermitent ringing comes from this line. > zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; > > Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) > > > The second point about CLIP is more difficult. First, let me indicate that I?m residing in France and therefore using carrier and phones complying to european standard. > About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. > > As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. > > I don?t know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel Seizure is present only in FSK-V23 Protocol. It > consists in a series of 90 to 300 alternate SPACE/MARK bits? ? > > Considering this, I guess that I should use the else branch in the test ? zchan->token_count > 1 ?. What does this token_count refer to and where should I set it. > > I?m still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. > > I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? > > Thanks in advance > > Fran?ois > > _______________________________________________ > 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/20100118/c5970d37/attachment-0001.html From anthony.minessale at gmail.com Mon Jan 18 14:24:21 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 18 Jan 2010 16:24:21 -0600 Subject: [Freeswitch-dev] openzap information + mod_say_fr contribution In-Reply-To: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> References: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> Message-ID: <191c3a031001181424v76c0b06ax1364f8ca79794601@mail.gmail.com> Hi, Thanks for the assistance. jira.freeswitch.org is where you submit code ring_on_ms is currently hard coded in ozmod_wanpipe.c:1069 We anticipate problems with CLIP in non-US lines since I do not have any to work with. We can work together with Sangoma to get this working right in the future once we have test lab to reproduce it. On Mon, Jan 18, 2010 at 4:01 PM, wrote: > Hello, > > > > First let me congratulate for this great software. > > > > I was trying to diagnose a problem with my analog sangoma card, and FXS > port giving intermitent ringing and not sending CLIP. > > > > I could see the intermitent ringing comes from this line. > > zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; > > > > Setting zchan->ring_time could solve my problem, so could anybody indicate > how to configure ring_on_ms (preferably in an XML configuration) > > > > > > The second point about CLIP is more difficult. First, let me indicate that > I?m residing in France and therefore using carrier and phones complying to > european standard. > > About CLIP, I could see the V23 standard modulating frequencies are being > populated (in fsk_modem_definitions in fsk.c). However, in > zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is > hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by > what I could read seem to apply to me), I could see no difference. > > > > As I could not find as much relevant information as I would like in V23 > (apart from the frequencies for the FSK), I was wondering what were I could > find some information that could help to fix this. > > > > I don?t know exactly what carrier_bits_start, carrier_bits_stop and > chan_sieze_bits refer to, but I could find in that document ( > http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel > Seizure is present only in FSK-V23 Protocol. It > > consists in a series of 90 to 300 alternate SPACE/MARK bits? ? > > > > Considering this, I guess that I should use the else branch in the test > ? zchan->token_count > 1 ?. What does this token_count refer to and where > should I set it. > > > > I?m still trying to figure this out (and will of course contribute the > result of all this), so any help appreciated. > > > > I also did a contribution to mod_say_fr (to take into account the specific > french idioms in saying date and time). Could you please tell me how to > contribute this ? > > > > Thanks in advance > > > > Fran?ois > > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100118/fa63d584/attachment.html From devel at thom.fr.eu.org Mon Jan 18 15:09:20 2010 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Tue, 19 Jan 2010 00:09:20 +0100 Subject: [Freeswitch-dev] openzap information + mod_say_fr contribution In-Reply-To: References: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> Message-ID: <000c01ca9893$44466900$ccd33b00$@fr.eu.org> What do you mean by ? raw recording ? ? How do I obtain this ? About mod_say_fr, shall I ? create a new issue ? ? Fran?ois De : freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] De la part de Michael Jerris Envoy? : lundi 18 janvier 2010 23:18 ? : freeswitch-dev at lists.freeswitch.org Objet : Re: [Freeswitch-dev] openzap information + mod_say_fr contribution In regards to the CLIP. The bulk of this code for V23 as you can see is already there, we just need this troubleshooting as we have never tested it. If you can get a raw recording of the channel it will probably help in testing. As for the mod_say_fr, please file this on http://jira.freeswitch.org Mike On Jan 18, 2010, at 5:01 PM, wrote: Hello, First let me congratulate for this great software. I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. I could see the intermitent ringing comes from this line. zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) The second point about CLIP is more difficult. First, let me indicate that I?m residing in France and therefore using carrier and phones complying to european standard. About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. I don?t know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel Seizure is present only in FSK-V23 Protocol. It consists in a series of 90 to 300 alternate SPACE/MARK bits ? Considering this, I guess that I should use the else branch in the test ? zchan->token_count > 1 ?. What does this token_count refer to and where should I set it. I?m still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? Thanks in advance Fran?ois _______________________________________________ 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/20100119/47416f4c/attachment.html From devel at thom.fr.eu.org Mon Jan 18 15:15:51 2010 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Tue, 19 Jan 2010 00:15:51 +0100 Subject: [Freeswitch-dev] openzap information + mod_say_fr contribution In-Reply-To: <191c3a031001181424v76c0b06ax1364f8ca79794601@mail.gmail.com> References: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> <191c3a031001181424v76c0b06ax1364f8ca79794601@mail.gmail.com> Message-ID: <001a01ca9894$2dbf8b70$893ea250$@fr.eu.org> I?d be glad to be that test lab if that is the meaning of your message. About ring_on_ms, changin this value may not be a good fix. I try to solve my problems in an empiric way. I found out that setting ring_time to -1 could solve my intermittent ringing problem, but that may cause problems with CLIP then (I took time to find the right document where CLIP FSK is defined). I don?t know how it is implemented in openzap (which TAS is used ). I?d be really happy to help, but I will need some pointers to start. Thanks Fran?ois De : freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] De la part de Anthony Minessale Envoy? : lundi 18 janvier 2010 23:24 ? : freeswitch-dev at lists.freeswitch.org Objet : Re: [Freeswitch-dev] openzap information + mod_say_fr contribution Hi, Thanks for the assistance. jira.freeswitch.org is where you submit code ring_on_ms is currently hard coded in ozmod_wanpipe.c:1069 We anticipate problems with CLIP in non-US lines since I do not have any to work with. We can work together with Sangoma to get this working right in the future once we have test lab to reproduce it. On Mon, Jan 18, 2010 at 4:01 PM, wrote: Hello, First let me congratulate for this great software. I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. I could see the intermitent ringing comes from this line. zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) The second point about CLIP is more difficult. First, let me indicate that I?m residing in France and therefore using carrier and phones complying to european standard. About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. I don?t know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel Seizure is present only in FSK-V23 Protocol. It consists in a series of 90 to 300 alternate SPACE/MARK bits ? Considering this, I guess that I should use the else branch in the test ? zchan->token_count > 1 ?. What does this token_count refer to and where should I set it. I?m still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? Thanks in advance Fran?ois _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100119/18edf9f8/attachment-0001.html From dujinfang at gmail.com Mon Jan 18 18:12:19 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 19 Jan 2010 10:12:19 +0800 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: Message-ID: <23f91031001181812h2285c520vcbd73b6e18625fa8@mail.gmail.com> Just an idea: mark each gateway with a digit so people can chose gateway by prefix with #1 #2 etc. And, better to make the digit setable so one may like #gw1..... >> >> Is the first provider assumed to be the default gateway? > > Right now, yes, but I am going to let you select tonight using the right > click mouse button on the account list on the bottom left corner of the main > window. > From jmesquita at freeswitch.org Mon Jan 18 20:09:21 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 19 Jan 2010 02:09:21 -0200 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: <23f91031001181812h2285c520vcbd73b6e18625fa8@mail.gmail.com> References: <23f91031001181812h2285c520vcbd73b6e18625fa8@mail.gmail.com> Message-ID: Ok, time to update and try again. Paul, the custom parameters bug is now fixed. Seven, take a look how it is now. I think that it might fit most of the users needs since we are able to assign shortcuts to actions (ALT+NUM for instance). Remember, right click on the account row on the Accounts table and check the new context menu. I will keep refactoring the code to add more features. FSComm is getting big so some organization is needed. Regards, Jo?o Mesquita FSComm Developer On Tue, Jan 19, 2010 at 12:12 AM, Seven Du wrote: > Just an idea: > > mark each gateway with a digit so people can chose gateway by prefix > with #1 #2 etc. And, better to make the digit setable so one may like > #gw1..... > > >> > >> Is the first provider assumed to be the default gateway? > > > > Right now, yes, but I am going to let you select tonight using the right > > click mouse button on the account list on the bottom left corner of the > main > > window. > > > > _______________________________________________ > 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/20100119/bf637705/attachment.html From ste.bossi at gmail.com Tue Jan 19 06:04:36 2010 From: ste.bossi at gmail.com (stefano bossi) Date: Tue, 19 Jan 2010 15:04:36 +0100 Subject: [Freeswitch-dev] thesis on live sip calls migration In-Reply-To: <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com> <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> Message-ID: <8ab521c31001190604n600f68a7h6f895eb6ea88acf9@mail.gmail.com> Hi Anthony, your last commit on sofia.c adds a very helpful function to extract sip_to_tag and sip_from_tag.. I was working on something similar but I also read the Cseq.. you can do this with these 3 lines of code if (sip->sip_cseq && sip->sip_cseq->cs_seq) { const char *sip_cseq = switch_core_session_sprintf(session,"%d", sip->sip_cseq->cs_seq); switch_channel_set_variable(channel, "sip_cseq", sip_cseq); } (It compiles, I hope it is correct too :) This cseq can be used for calls failover in the most part of cases, I only wonder if this is still true after any other request during dialog than INVITE. bye On Mon, Jan 4, 2010 at 5:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Hi, > When would you like to continue this conversation? > > > On Fri, Dec 18, 2009 at 4:38 AM, stefano bossi wrote: > >> Hi all, >> >> as promised in #freeswitch-dev I'll try to explain better what I've done >> and I would like to achieve for my thesis. >> >> >> At the moment I successfully patched sofsip_cli and now it can do >> something near active call migration :) >> Thanks to the versatility of nua api I could achieve my objective with >> little effort. >> Basically I call someone, then I kill sofsip_cli and I restart it. I >> launch my "reinvite" command and the call can continue normally. The called >> client doesn't realize what happened. >> In this case I write call data in a file and when I launch the reinvite I >> read directly from this file. >> Sofsip_cli sends an INVITE message with some TAG appended to the >> nua_handle. So the called phone thinks to receive an in-session invite and >> simply refreshes the call, but Sofsip_cli instead allocates all the stuff >> for a new call. In this way I can avoid to modify directly the RTP part. It >> seems to be all simpler. >> >> I even wrote a little module for FS able to monitor(using the SIP OPTION >> message) the life of a specific sofia profile.. It's just a proof of concept >> but it can do failover(without live call migration) between 2 machines in >> about 50ms. >> This is possible with these actions: >> >> - registrations are shared with odbc >> >> >> - on the start of the module(present only on the backup machine): >> >> >> 1. I set an arp rule blocking the arp response for the virtual IP (set >> on the primary machine) >> 2. I set the virtual IP on the backup machine( but no one knows thanks >> to the arp rule and so I can bind to vIP) >> 3. I prepare and run the sip profile (on the backup)... loading here >> the profile permits to save a lot of time during reaction >> >> >> - on the reaction >> >> >> 1. I remove the arp rule >> 2. I send a gratuitous arp request >> >> As I said in about 50ms sip clients can call again. Maybe this time can >> decrease using NETLINK socket for the arp table. >> >> The union of these 2 works will give us a very fast "live profile >> migration" :D >> >> There some points to discuss: >> >> - switch_core_session_resurrect_channel: I didn't know this function, >> I need to understand what it offers, maybe tomorrow ;) >> - the propagation of call state variables: my first idea was to use >> the multicast events adding some headers to send all the necessary data (but >> anthm proposed XML) >> - I thought only to sofia aspects.. >> >> In next days I'll try to understand where to hook in FS to try these >> ideas. When I'll have something ready (and without very very big errors:) >> I'll be happy to send you the patch. >> This is my first work on something real like FS.. I'm opened to any kind >> of suggestions!* >> >> *please contact me for further clarifications >> >> Thanks >> >> Stefano Bossi >> >> >> >> _______________________________________________ >> 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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/20100119/cc7faf1a/attachment.html From ste.bossi at gmail.com Tue Jan 19 06:04:45 2010 From: ste.bossi at gmail.com (stefano bossi) Date: Tue, 19 Jan 2010 15:04:45 +0100 Subject: [Freeswitch-dev] thesis on live sip calls migration In-Reply-To: <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com> <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> Message-ID: <8ab521c31001190604g6a933f21n159946ae31852d63@mail.gmail.com> Hi Anthony, your last commit on sofia.c adds a very helpful function to extract sip_to_tag and sip_from_tag.. I was working on something similar but I also read the Cseq.. you can do this with these 3 lines of code if (sip->sip_cseq && sip->sip_cseq->cs_seq) { const char *sip_cseq = switch_core_session_sprintf(session,"%d", sip->sip_cseq->cs_seq); switch_channel_set_variable(channel, "sip_cseq", sip_cseq); } (It compiles, I hope it is correct too :) This cseq can be used for calls failover in the most part of cases, I only wonder if this is still true after any other request during dialog than INVITE. bye On Mon, Jan 4, 2010 at 5:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Hi, > When would you like to continue this conversation? > > > On Fri, Dec 18, 2009 at 4:38 AM, stefano bossi wrote: > >> Hi all, >> >> as promised in #freeswitch-dev I'll try to explain better what I've done >> and I would like to achieve for my thesis. >> >> >> At the moment I successfully patched sofsip_cli and now it can do >> something near active call migration :) >> Thanks to the versatility of nua api I could achieve my objective with >> little effort. >> Basically I call someone, then I kill sofsip_cli and I restart it. I >> launch my "reinvite" command and the call can continue normally. The called >> client doesn't realize what happened. >> In this case I write call data in a file and when I launch the reinvite I >> read directly from this file. >> Sofsip_cli sends an INVITE message with some TAG appended to the >> nua_handle. So the called phone thinks to receive an in-session invite and >> simply refreshes the call, but Sofsip_cli instead allocates all the stuff >> for a new call. In this way I can avoid to modify directly the RTP part. It >> seems to be all simpler. >> >> I even wrote a little module for FS able to monitor(using the SIP OPTION >> message) the life of a specific sofia profile.. It's just a proof of concept >> but it can do failover(without live call migration) between 2 machines in >> about 50ms. >> This is possible with these actions: >> >> - registrations are shared with odbc >> >> >> - on the start of the module(present only on the backup machine): >> >> >> 1. I set an arp rule blocking the arp response for the virtual IP (set >> on the primary machine) >> 2. I set the virtual IP on the backup machine( but no one knows thanks >> to the arp rule and so I can bind to vIP) >> 3. I prepare and run the sip profile (on the backup)... loading here >> the profile permits to save a lot of time during reaction >> >> >> - on the reaction >> >> >> 1. I remove the arp rule >> 2. I send a gratuitous arp request >> >> As I said in about 50ms sip clients can call again. Maybe this time can >> decrease using NETLINK socket for the arp table. >> >> The union of these 2 works will give us a very fast "live profile >> migration" :D >> >> There some points to discuss: >> >> - switch_core_session_resurrect_channel: I didn't know this function, >> I need to understand what it offers, maybe tomorrow ;) >> - the propagation of call state variables: my first idea was to use >> the multicast events adding some headers to send all the necessary data (but >> anthm proposed XML) >> - I thought only to sofia aspects.. >> >> In next days I'll try to understand where to hook in FS to try these >> ideas. When I'll have something ready (and without very very big errors:) >> I'll be happy to send you the patch. >> This is my first work on something real like FS.. I'm opened to any kind >> of suggestions!* >> >> *please contact me for further clarifications >> >> Thanks >> >> Stefano Bossi >> >> >> >> _______________________________________________ >> 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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/20100119/0c06f10f/attachment-0001.html From sipx95 at eljv.com Tue Jan 19 07:54:04 2010 From: sipx95 at eljv.com (Eric Varsanyi) Date: Tue, 19 Jan 2010 09:54:04 -0600 Subject: [Freeswitch-dev] Autoconfig problem on FC12 Message-ID: I'm building freeswitch RPM packages using the sipxecs build system and the rpm packaging step was failing because libjs.so was not being built. After some digging it looks to me like a dependency problem in the libjs autoconfig setup. The patch below works around the issue but there's likely a better way. -Eric Varsanyi Details: The proximal issue is that EGREP is not defined when checking for the ability to make shared libs. The problem is some sort of dependency issue between AC_CHECK_HEADER and AC_PROG_LIBTOOL. It appears that using AC_CHECK_HEADER at 'autoconf' time remembers that the check for, say, EGREP has been done and prevents it from being issued again in the resulting configure script, however if AC_CHECK_HEADER is used in a runtime 'if' block (the 'if' runs when configure runs, not autoconf) and that if does not pay off AC_PROG_LIBTOOL ends up w/o EGREP defined and the check if we can build a shared library fails (and thus no .so is built). A simple workaround is to move AC_PROG_LIBTOOL up above the AC_ARG_WITH([dso ... check which uses AC_CHECK_HEADER in a bash if block (which does not pay off on linux). I'm no autoconf wizard and my lame attempts to force it to check for EGREP again failed -- it may be you're just not supposed to use some things inside an 'if' block at 'configure' time. Index: libs/js/configure.ac =================================================================== --- libs/js/configure.ac (revision 16398) +++ libs/js/configure.ac (working copy) @@ -154,6 +154,8 @@ CPPFLAGS="$CPPFLAGS -DJS_HAS_XML_SUPPORT=1" fi +AC_PROG_LIBTOOL + dnl # configure option --with-dso AC_ARG_WITH([dso], AS_HELP_STRING([--with-dso], [build without DSO object (allows run-time process extending)]), @@ -229,8 +231,6 @@ with_tags="" -AC_PROG_LIBTOOL - AC_CONFIG_HEADERS(config.h) AC_CONFIG_FILES([Makefile js.pc js-config.sh src/perlconnect/Makefile.PL]) AC_OUTPUT From Suneel.Papineni at mettoni.com Tue Jan 19 09:14:51 2010 From: Suneel.Papineni at mettoni.com (Suneel Papineni) Date: Tue, 19 Jan 2010 17:14:51 -0000 Subject: [Freeswitch-dev] Understansing of mod_flite Config Message-ID: <3181A30B8C35AB4AA8577B78DDF461380653E3CE@nickel.mettonigroup.com> Hi, I am trying to test mod_flite to announce a message like "You are 5th in Queue". mod_flite is compiled properly. I am using Windows OS and event sockets to send the commands. A dial-plan is written to park all calls that are coming to Freeswitch. After call is parked I am sending message to play the announcement. sendmsg [callID] call-command: execute execute-app-name: speak execute-app-arg: 'Hi you are 5th in Queue' This is giving an error in FS as: 2010-01-19 15:40:37.814740 [ERR] mod_dptools.c:1505 Invalid Params! [NULL][NULL][Hi you are 5th in Queue] I tried to debug this, but failed to find the issue. It seems as one argument is passed, engine and voice are null, so this error is thrown ( I assume ). Tried to send message like "execute-app-arg: 'flite'|'david'|'Hi you are 5th in Queue'". This took as 3 arguments but still thrown an error saying "Invalid speech module(flite)". Could you please let me know where I am going wrong. Also where do I need to configure engine and voice parameters, so that system could pick up automatically. Tried to set these parameters in conference.conf.xml, but not worked. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/55022048/attachment.html From mike at jerris.com Tue Jan 19 09:29:29 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 19 Jan 2010 12:29:29 -0500 Subject: [Freeswitch-dev] Understansing of mod_flite Config In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380653E3CE@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF461380653E3CE@nickel.mettonigroup.com> Message-ID: <7A159758-C9DF-4129-894A-2FB3BF72F432@jerris.com> On Jan 19, 2010, at 12:14 PM, Suneel Papineni wrote: > Hi, > > I am trying to test mod_flite to announce a message like "You are 5th in Queue". mod_flite is compiled properly. I am using Windows OS and event sockets to send the commands. A dial-plan is written to park all calls that are coming to Freeswitch. After call is parked I am sending message to play the announcement. > > sendmsg [callID] > call-command: execute > execute-app-name: speak > execute-app-arg: 'Hi you are 5th in Queue' > > This is giving an error in FS as: > 2010-01-19 15:40:37.814740 [ERR] mod_dptools.c:1505 Invalid Params! [NULL][NULL][Hi you are 5th in Queue] > > I tried to debug this, but failed to find the issue. It seems as one argument is passed, engine and voice are null, so this error is thrown ( I assume ). Tried to send message like ?execute-app-arg: 'flite'|'david'|'Hi you are 5th in Queue'?. This took as 3 arguments but still thrown an error saying "Invalid speech module(flite)". This looks more correct. But mod_flite is not loaded. > > Could you please let me know where I am going wrong. Also where do I need to configure engine and voice parameters, so that system could pick up automatically. Tried to set these parameters in conference.conf.xml, but not worked. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/687cd1cb/attachment.html From brian at freeswitch.org Tue Jan 19 09:36:36 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 19 Jan 2010 11:36:36 -0600 Subject: [Freeswitch-dev] Changes for deb and rpm files. Message-ID: <5CA0D17B-6BD4-4BC6-B7F9-4A117A896C86@freeswitch.org> Heads up. http://jira.freeswitch.org/browse/FSCONFIG-17 Someone will need to update the configs for the debs and rpm's /b From mike at jerris.com Tue Jan 19 10:14:35 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 19 Jan 2010 13:14:35 -0500 Subject: [Freeswitch-dev] Autoconfig problem on FC12 In-Reply-To: References: Message-ID: <4DB49074-2CD0-4E6D-BF48-0C76C6218ED9@jerris.com> I fixed this in tree basically with your fix below. Basically these checks should always be done, not just in certain conditions. The problem with this is they have error messages right in the tags, so it will take a bit of rework that is probably not worth it. Mike On Jan 19, 2010, at 10:54 AM, Eric Varsanyi wrote: > I'm building freeswitch RPM packages using the sipxecs build system and the rpm packaging step was failing because libjs.so was not being built. After some digging it looks to me like a dependency problem in the libjs autoconfig setup. The patch below works around the issue but there's likely a better way. > > -Eric Varsanyi > > Details: > > The proximal issue is that EGREP is not defined when checking for the ability to make shared libs. > > The problem is some sort of dependency issue between AC_CHECK_HEADER and AC_PROG_LIBTOOL. > > It appears that using AC_CHECK_HEADER at 'autoconf' time remembers that the check for, say, EGREP has been done and prevents it from being issued again in the resulting configure script, however if AC_CHECK_HEADER is used in a runtime 'if' block (the 'if' runs when configure runs, not autoconf) and that if does not pay off AC_PROG_LIBTOOL ends up w/o EGREP defined and the check if we can build a shared library fails (and thus no .so is built). > > A simple workaround is to move AC_PROG_LIBTOOL up above the AC_ARG_WITH([dso ... check which uses AC_CHECK_HEADER in a bash if block (which does not pay off on linux). I'm no autoconf wizard and my lame attempts to force it to check for EGREP again failed -- it may be you're just not supposed to use some things inside an 'if' block at 'configure' time. > > Index: libs/js/configure.ac > =================================================================== > --- libs/js/configure.ac (revision 16398) > +++ libs/js/configure.ac (working copy) > @@ -154,6 +154,8 @@ > CPPFLAGS="$CPPFLAGS -DJS_HAS_XML_SUPPORT=1" > fi > > +AC_PROG_LIBTOOL > + > dnl # configure option --with-dso > AC_ARG_WITH([dso], > AS_HELP_STRING([--with-dso], [build without DSO object (allows run-time process extending)]), > @@ -229,8 +231,6 @@ > > with_tags="" > > -AC_PROG_LIBTOOL > - > AC_CONFIG_HEADERS(config.h) > AC_CONFIG_FILES([Makefile js.pc js-config.sh src/perlconnect/Makefile.PL]) > AC_OUTPUT > > > > _______________________________________________ > 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 devel at thom.fr.eu.org Tue Jan 19 11:09:15 2010 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Tue, 19 Jan 2010 20:09:15 +0100 Subject: [Freeswitch-dev] Openzap and intermitent ringing Message-ID: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> So I finally could find a fix. Then problem is the ring_on and ring_off length as defined by wp_globals in ozmod_wanpipe.c For France (and I guess all other European countries), the ring on and off length shall be 1700ms and 3300ms. That correspond (at least for Europe) to the length of the ringing tone. So I was wondering I it would not be a good idea to take these values from the tones.conf file. I wonder if that is possible in wp_open_range function in ozmod_wanpipe.c. Could anybody indicate if there is any diagram of how each module in openzap calls the other, then that could be a good starting point (I don?t really get the interactions/calls between ozmod_wanpipe.c and ozmod_analog.c). Also, I would need to know if this tone generation should apply to span types other than FXS. Thanks Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/d70a49c2/attachment.html From devel at thom.fr.eu.org Tue Jan 19 11:15:04 2010 From: devel at thom.fr.eu.org (devel at thom.fr.eu.org) Date: Tue, 19 Jan 2010 20:15:04 +0100 Subject: [Freeswitch-dev] openzap information In-Reply-To: References: <002301ca9889$c454be30$4cfe3a90$@fr.eu.org> Message-ID: <002001ca993b$b49f2810$1ddd7830$@fr.eu.org> Could you please give me some information as starting point on this. I don?t know for BELL standard, but in V23, there different ways of sending the CLIP : before the ringing signal and during the first silence between 2 consecutive ring signals. By looking at the sequence, it seems to me as if the first one (CLIP before the ringing signal) is used. Again, I don?t know in the BELL standard (but I guess it should be somehow similar), the need to be some signal (called TAS) sent to the TE to let him know the CLIP is coming (so that it can set the line impedance in order to receive it). So could you confirm that CLIP before the ringing signal is coded in openzap, and indicate which type of TAS is used and where is this part coded in openzap. Thanks in advance Fran?ois De : freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] De la part de Michael Jerris Envoy? : lundi 18 janvier 2010 23:18 ? : freeswitch-dev at lists.freeswitch.org Objet : Re: [Freeswitch-dev] openzap information + mod_say_fr contribution In regards to the CLIP. The bulk of this code for V23 as you can see is already there, we just need this troubleshooting as we have never tested it. If you can get a raw recording of the channel it will probably help in testing. As for the mod_say_fr, please file this on http://jira.freeswitch.org Mike On Jan 18, 2010, at 5:01 PM, wrote: Hello, First let me congratulate for this great software. I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. I could see the intermitent ringing comes from this line. zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) The second point about CLIP is more difficult. First, let me indicate that I?m residing in France and therefore using carrier and phones complying to european standard. About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. I don?t know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that ? The Channel Seizure is present only in FSK-V23 Protocol. It consists in a series of 90 to 300 alternate SPACE/MARK bits ? Considering this, I guess that I should use the else branch in the test ? zchan->token_count > 1 ?. What does this token_count refer to and where should I set it. I?m still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? Thanks in advance Fran?ois _______________________________________________ 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/20100119/6d2f794a/attachment.html From anthony.minessale at gmail.com Tue Jan 19 11:22:45 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Jan 2010 13:22:45 -0600 Subject: [Freeswitch-dev] Openzap and intermitent ringing In-Reply-To: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> References: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> Message-ID: <191c3a031001191122i268f7079m3324abe482e2e77f@mail.gmail.com> There is an openzap irc channel you could come to and talk to people about it #openzap on irc.freenode.net On Tue, Jan 19, 2010 at 1:09 PM, wrote: > So I finally could find a fix. > > > > Then problem is the ring_on and ring_off length as defined by wp_globals in > ozmod_wanpipe.c > > For France (and I guess all other European countries), the ring on and off > length shall be 1700ms and 3300ms. > > That correspond (at least for Europe) to the length of the ringing tone. > > So I was wondering I it would not be a good idea to take these values from > the tones.conf file. > > > > I wonder if that is possible in wp_open_range function in ozmod_wanpipe.c. > > Could anybody indicate if there is any diagram of how each module in > openzap calls the other, then that could be a good starting point (I don?t > really get the interactions/calls between ozmod_wanpipe.c and > ozmod_analog.c). Also, I would need to know if this tone generation should > apply to span types other than FXS. > > > > Thanks > > > > Fran?ois > > > > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100119/02f779a7/attachment-0001.html From moises.silva at gmail.com Tue Jan 19 12:00:14 2010 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 19 Jan 2010 15:00:14 -0500 Subject: [Freeswitch-dev] Openzap and intermitent ringing In-Reply-To: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> References: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> Message-ID: Hello Fran?ois, On Tue, Jan 19, 2010 at 2:09 PM, wrote: > So I finally could find a fix. > > > > Then problem is the ring_on and ring_off length as defined by wp_globals in > ozmod_wanpipe.c > > For France (and I guess all other European countries), the ring on and off > length shall be 1700ms and 3300ms. > > That correspond (at least for Europe) to the length of the ringing tone. > > So I was wondering I it would not be a good idea to take these values from > the tones.conf file. > > I just committed the fix to configure ring on / off from the configuration file (wanpipe.conf), given that these values are wanpipe specific. There are other wp global values already taken from the configuration, see wanpipe_init() in the same file. > Could anybody indicate if there is any diagram of how each module in > openzap calls the other, then that could be a good starting point (I don?t > really get the interactions/calls between ozmod_wanpipe.c and > ozmod_analog.c). > This diagram is all we have AFAIK: http://wiki.freeswitch.org/wiki/OpenZAP OpenZAP ( which is being renamed to FreeTDM as we speak) provides abstraction to both signaling and board I/O APIs. In short, you have therefore 2 type of modules, signaling and I/O modules. I/O modules are the low level modules that use the driver-specific API to talk to the board, either Sangoma, PIKA or Zaptel-based boards. Therefore, the use can execute I/O commands using openzap API and does not have to care about which type of board is talking to. For example, ozmod_analog does this: zap_channel_command(zchan, ZAP_COMMAND_GENERATE_RING_ON), the openzap core will then check which type of I/O module is associated to the channel (wanpipe, zt, pika etc) and ask the IO module to execute the command using the specific API for the board. The signaling module is supposed to abstract the telephony signaling (PRI, R2, FXO/FXS, SS7 etc). So, if a user wants to place a call, regardless of the signaling, it will only issue zap_channel_open() and then zap_channel_outgoing_call() which will take care of checking which signaling module is associated to the span where that channel belongs and ask the signaling module to place the call. There are some I/O features that may not be available in some drivers (like flash in wanpipe) and are therefore emulated by the software layer of the I/O wanpipe module. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/2933d717/attachment.html From msc at freeswitch.org Tue Jan 19 15:21:55 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Jan 2010 15:21:55 -0800 Subject: [Freeswitch-dev] NOTICE: mod_iax slated for unsupported on Feb. 5 Message-ID: <87f2f3b91001191521y3aec5aadp6ebee73945bb7e0c@mail.gmail.com> To everyone in the FreeSWITCH community: We would like to put a call out and let everyone know that unless we find someone who wants to be the maintainer for mod_iax it will be moved to unsupported status as of Friday February 5th. Anyone who wishes to maintain the module will need to be in charge of all bug reports, updates, patches, and technical support questions. If are in a position to maintain mod_iax then please contact Brian West offlist: brian at freeswitch.org. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/f8ce967c/attachment.html From tonhudung at gmail.com Tue Jan 19 22:04:22 2010 From: tonhudung at gmail.com (Alex To) Date: Wed, 20 Jan 2010 14:04:22 +0800 Subject: [Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in) Message-ID: <000001ca9996$73352f90$599f8eb0$@com> Hi, Could somebody shed some lights on how the mentioned method works ? What does it do exactly ? My best guess is it takes in a channel, a string and does something to the string (escape the string ? split the string ? or something like that? But why need a channel parameter?) If there is already an explanation somewhere about this, please point it out for me or possibly just tell me what configuration should I make to trigger this method to be called. I can use debugger and figure it out by myself. Any help would be greatly appreciated. Thanks very much Regards Alex To -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100120/dd04f87f/attachment.html From vipkilla at gmail.com Wed Jan 20 06:55:08 2010 From: vipkilla at gmail.com (vip killa) Date: Wed, 20 Jan 2010 09:55:08 -0500 Subject: [Freeswitch-dev] fscomm with celt Message-ID: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> has anyone built fscomm with the celt codec for windows? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100120/eb3a8d2a/attachment.html From rupa at rupa.com Wed Jan 20 07:29:48 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 20 Jan 2010 09:29:48 -0600 Subject: [Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in) In-Reply-To: <000001ca9996$73352f90$599f8eb0$@com> References: <000001ca9996$73352f90$599f8eb0$@com> Message-ID: It expands channel vars in the string. On Wed, Jan 20, 2010 at 12:04 AM, Alex To wrote: > Hi, > > > > Could somebody shed some lights on how the mentioned method works ? What > does it do exactly ? My best guess is it takes in a channel, a string and > does something to the string (escape the string ? split the string ? or > something like that? But why need a channel parameter?) > > > > If there is already an explanation somewhere about this, please point it > out for me or possibly just tell me what configuration should I make to > trigger this method to be called. I can use debugger and figure it out by > myself. > > > > Any help would be greatly appreciated. > > > > Thanks very much > > > > Regards > > > > Alex To > > _______________________________________________ > 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100120/e275787a/attachment.html From anthony.minessale at gmail.com Wed Jan 20 10:48:35 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 20 Jan 2010 12:48:35 -0600 Subject: [Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in) In-Reply-To: <000001ca9996$73352f90$599f8eb0$@com> References: <000001ca9996$73352f90$599f8eb0$@com> Message-ID: <191c3a031001201048n53c6c542tdd1a0f20b5c7c03e@mail.gmail.com> say you had 2 vars test1=foo and test2=bar it would turn the string "${test1} ${test2}" into "foo bar" if the returned string does not return the exact pointer that was input, it must be freed. On Wed, Jan 20, 2010 at 12:04 AM, Alex To wrote: > Hi, > > > > Could somebody shed some lights on how the mentioned method works ? What > does it do exactly ? My best guess is it takes in a channel, a string and > does something to the string (escape the string ? split the string ? or > something like that? But why need a channel parameter?) > > > > If there is already an explanation somewhere about this, please point it > out for me or possibly just tell me what configuration should I make to > trigger this method to be called. I can use debugger and figure it out by > myself. > > > > Any help would be greatly appreciated. > > > > Thanks very much > > > > Regards > > > > Alex To > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100120/a728026f/attachment-0001.html From moises.silva at gmail.com Wed Jan 20 11:01:03 2010 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 20 Jan 2010 14:01:03 -0500 Subject: [Freeswitch-dev] Openzap and intermitent ringing In-Reply-To: <2767a9f20c8cc35f4c1c62c43528f9b2@thom.fr.eu.org> References: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> <2767a9f20c8cc35f4c1c62c43528f9b2@thom.fr.eu.org> Message-ID: Hi again Fran?ois, Please keep the conversation in the freeswitch-dev mailing list, unless you want private paid support ;-) See the answer to your question below ... On Wed, Jan 20, 2010 at 3:57 AM, Fran?ois Legal wrote: > Do you know if there is a way (except using an oscilloscope to sample the > line signal) to get a dump of the data sent to the card by openzap (is it > samples or something) and vice versa. > > > > What I would like to do is see what is returned to openzap when a call > arrives on an FXO port (CLIP is working in that direction) and compare it to > what is being sent to the card when sending CLIP to and FXS port. > Once the call gets into the computer we're talking about digital audio samples. In a typical configuration a Sangoma board will give you 160 samples of audio each 20ms (most likely encoded in ulaw, that can be verified in /etc/wanpipe/wanpipex.conf for the FXO/FXS board, look at the TDMV_LAW parameter). There is an openzap API command to record the digital audio read from the board, check ZAP_COMMAND_TRACE_INPUT and ZAP_COMMAND_TRACE_OUTPUT. What we currently lack is a FreeSWITCH command in mod_openzap to enable/disable the audio tracing. It should be trivial to hack mod_openzap.c and add that command, if you need help with that ping me on IRC (moy). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100120/ea00cb75/attachment.html From mardy at voysys.com Wed Jan 20 15:14:48 2010 From: mardy at voysys.com (Mardy Marshall) Date: Wed, 20 Jan 2010 18:14:48 -0500 Subject: [Freeswitch-dev] Problem with event_socket application when serving as a consultative transfer target Message-ID: I'm having a problem with an event_socket application which is the transfer target of a consultative transfer. When the transferor begins the transfer process, the outbound event_socket session is created and everything looks normal. When the transfer is completed, switch_ivr_bridge fires a CHANNEL_EXECUTE / "uuid_bridge" event, supplying a new UUID for the "Other-Leg" which corresponds to the transferee. From this point on, I no longer receive events specific to this new UUID, such as DTMF events or CHANNEL_EXECUTE_COMPLETE events after issuing "playback" commands. Regarding the "playback" command, if I direct the command to the "Other-Leg" UUID, I get audio, but no events when it is done. On the other hand, if I direct the command to the original UUID, the one corresponding to the transferrer call leg, I get no audio, but do get completion events. I have tried re-issuing a "myevents" command using the new UUID, but the command fails due to it only being allowed to be called once per session. Am I missing something in order to handle the transfer or have I run into a bug? Thanks, -Mardy From anthony.minessale at gmail.com Wed Jan 20 16:11:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 20 Jan 2010 18:11:27 -0600 Subject: [Freeswitch-dev] Problem with event_socket application when serving as a consultative transfer target In-Reply-To: References: Message-ID: <191c3a031001201611k7c8a5029u26e80f4eb1b75f2b@mail.gmail.com> you could try using the filter command to filter on the various uuid rather than locking it on to 1 channel with myevents essentially once it's transferred your leg is dead so you can stick around on the socket if you want but you can't control the other channel. If filter does not work we could look at a way to myevents over to another session but it may be tricky and involve a bounty. On Wed, Jan 20, 2010 at 5:14 PM, Mardy Marshall wrote: > I'm having a problem with an event_socket application which is the transfer > target of a consultative transfer. When the transferor begins the transfer > process, the outbound event_socket session is created and everything looks > normal. When the transfer is completed, switch_ivr_bridge fires a > CHANNEL_EXECUTE / "uuid_bridge" event, supplying a new UUID for the > "Other-Leg" which corresponds to the transferee. From this point on, I no > longer receive events specific to this new UUID, such as DTMF events or > CHANNEL_EXECUTE_COMPLETE events after issuing "playback" commands. > Regarding the "playback" command, if I direct the command to the > "Other-Leg" UUID, I get audio, but no events when it is done. On the other > hand, if I direct the command to the original UUID, the one corresponding to > the transferrer call leg, I get no audio, but do get completion events. I > have tried re-issuing a "myevents" command using the new UUID, but the > command fails due to it only being allowed to be called once per session. > > Am I missing something in order to handle the transfer or have I run into a > bug? > > Thanks, > > -Mardy > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100120/52d78a44/attachment.html From tonhudung at gmail.com Wed Jan 20 18:10:15 2010 From: tonhudung at gmail.com (Alex To) Date: Thu, 21 Jan 2010 10:10:15 +0800 Subject: [Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in) In-Reply-To: <191c3a031001201048n53c6c542tdd1a0f20b5c7c03e@mail.gmail.com> References: <000001ca9996$73352f90$599f8eb0$@com> <191c3a031001201048n53c6c542tdd1a0f20b5c7c03e@mail.gmail.com> Message-ID: <003301ca9a3e$e4302210$ac906630$@com> Thanks Anthony and Rupa, it's clear now. J Alex To From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, January 21, 2010 2:49 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in) say you had 2 vars test1=foo and test2=bar it would turn the string "${test1} ${test2}" into "foo bar" if the returned string does not return the exact pointer that was input, it must be freed. On Wed, Jan 20, 2010 at 12:04 AM, Alex To wrote: Hi, Could somebody shed some lights on how the mentioned method works ? What does it do exactly ? My best guess is it takes in a channel, a string and does something to the string (escape the string ? split the string ? or something like that? But why need a channel parameter?) If there is already an explanation somewhere about this, please point it out for me or possibly just tell me what configuration should I make to trigger this method to be called. I can use debugger and figure it out by myself. Any help would be greatly appreciated. Thanks very much Regards Alex To _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100121/6d1c3540/attachment-0001.html From carlos.talbot at gmail.com Tue Jan 19 15:08:06 2010 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Tue, 19 Jan 2010 17:08:06 -0600 Subject: [Freeswitch-dev] FSComm VS2008 Project file - updated In-Reply-To: References: Message-ID: <5800526b1001191508h4ec76580jda9452a00b2c68e0@mail.gmail.com> Joao, I was in the process of incorporating these changes but since svn 16399 I'm unable to link the executable. The new file account.cpp has some link errors: 1>Compiling... 1>account.cpp 1>Linking... 1>account.obj : error LNK2001: unresolved external symbol "public: virtual struct QMetaObject const * __thiscall Account::metaObject(void)const " (?metaObject at Account@@UBEPBUQMetaObject@@XZ) 1>account.obj : error LNK2001: unresolved external symbol "public: virtual void * __thiscall Account::qt_metacast(char const *)" (?qt_metacast at Account @@UAEPAXPBD at Z) 1>account.obj : error LNK2001: unresolved external symbol "public: virtual int __thiscall Account::qt_metacall(enum QMetaObject::Call,int,void * *)" (?qt_metacall at Account@@UAEHW4Call at QMetaObject@@HPAPAX at Z) 1>release\FSComm.exe : fatal error LNK1120: 3 unresolved externals 1>Build log was saved at "file://g:\freeswitch_dev\fscomm\release\BuildLog.htm" 1>FSComm - 4 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== regards, Carlos 2010/1/15 Jo?o Mesquita > Carlos, Jeff, > > Would any of you take a look at that for us, please? > > Paul, thank you for the contribution. Would you please, next time, post a > Jira so we are able to keep track of all contributions? > > Regards, > Jo?o Mesquita > FSComm Lead Developer > > > On Fri, Jan 15, 2010 at 10:26 PM, Paul Li wrote: > >> The present FSComm VS2008 Project file need to be updated as attached >> in order to build it with Visual Studio 2008 for the issues: >> >> (1) prefaccounts.h/prefaccounts.cpp need to be added to the project. >> (2) "Generated Files" need to be filtered properly for Debug/Release >> configuration, so that the proper files are used for the corresponding >> configuration. >> (3) Link to proper "FreeSwitchCore.lib", based on the selected >> configuration. >> (4) All intermediate files now go to $(IntDir), except for "ui_*.h". >> >> All these issues have been corrected with the attached .proj file. >> >> Thanks for sharing the app! >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100119/fd4fa943/attachment.html From carlos.talbot at gmail.com Wed Jan 20 10:16:33 2010 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 20 Jan 2010 12:16:33 -0600 Subject: [Freeswitch-dev] fscomm with celt In-Reply-To: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> References: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> Message-ID: <5800526b1001201016ubb2bd7r60caa1850277e6c9@mail.gmail.com> Until now that was not possible because there was no VS project file to build CELT on Windows I've just committed updates to resolve this. I've also recompiled FSComm with mod_celt and am uploading the most recent install file. On Wed, Jan 20, 2010 at 8:55 AM, vip killa wrote: > has anyone built fscomm with the celt codec for windows? > > _______________________________________________ > 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/20100120/377bf6a3/attachment.html From carlos.talbot at gmail.com Wed Jan 20 19:11:41 2010 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 20 Jan 2010 21:11:41 -0600 Subject: [Freeswitch-dev] fscomm with celt In-Reply-To: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> References: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> Message-ID: <5800526b1001201911o5112d4fby329f392d88ad77c7@mail.gmail.com> Until now that was not possible because there was no VS project file to build CELT on Windows I've just committed updates to resolve this. I've also recompiled FSComm with mod_celt and have uploaded the most recent install file. On Wed, Jan 20, 2010 at 8:55 AM, vip killa wrote: > has anyone built fscomm with the celt codec for windows? > > _______________________________________________ > 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/20100120/9cab190e/attachment.html From jmesquita at freeswitch.org Wed Jan 20 20:04:48 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 21 Jan 2010 02:04:48 -0200 Subject: [Freeswitch-dev] fscomm with celt In-Reply-To: <5800526b1001201016ubb2bd7r60caa1850277e6c9@mail.gmail.com> References: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> <5800526b1001201016ubb2bd7r60caa1850277e6c9@mail.gmail.com> Message-ID: Like usual, Carlos is tha man. Thanks Carlos! Regards, Jo?o Mesquita FSComm Developer On Wed, Jan 20, 2010 at 4:16 PM, Carlos Talbot wrote: > Until now that was not possible because there was no VS project file to > build CELT on Windows > > I've just committed updates to resolve this. I've also recompiled FSComm > with mod_celt and am uploading the most recent install file. > > On Wed, Jan 20, 2010 at 8:55 AM, vip killa wrote: > >> has anyone built fscomm with the celt codec for windows? >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100121/61a5df33/attachment.html From jmesquita at freeswitch.org Wed Jan 20 20:18:30 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 21 Jan 2010 02:18:30 -0200 Subject: [Freeswitch-dev] FSComm VS2008 Project file - updated In-Reply-To: <5800526b1001191508h4ec76580jda9452a00b2c68e0@mail.gmail.com> References: <5800526b1001191508h4ec76580jda9452a00b2c68e0@mail.gmail.com> Message-ID: Guys, this was already fixed. Thank you. Jo?o Mesquita FSComm Developer On Tue, Jan 19, 2010 at 9:08 PM, Carlos Talbot wrote: > Joao, > > I was in the process of incorporating these changes but since svn 16399 > I'm unable to link the executable. The new file account.cpp has some link > errors: > > 1>Compiling... > 1>account.cpp > 1>Linking... > 1>account.obj : error LNK2001: unresolved external symbol "public: virtual > struct QMetaObject const * __thiscall Account::metaObject(void)const " > (?metaObject at Account@@UBEPBUQMetaObject@@XZ) > 1>account.obj : error LNK2001: unresolved external symbol "public: virtual > void * __thiscall Account::qt_metacast(char const *)" (?qt_metacast at Account > @@UAEPAXPBD at Z) > 1>account.obj : error LNK2001: unresolved external symbol "public: virtual > int __thiscall Account::qt_metacall(enum QMetaObject::Call,int,void * *)" > (?qt_metacall at Account@@UAEHW4Call at QMetaObject@@HPAPAX at Z) > 1>release\FSComm.exe : fatal error LNK1120: 3 unresolved externals > 1>Build log was saved at > "file://g:\freeswitch_dev\fscomm\release\BuildLog.htm" > 1>FSComm - 4 error(s), 0 warning(s) > ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== > > regards, > > Carlos > > 2010/1/15 Jo?o Mesquita > > Carlos, Jeff, >> >> Would any of you take a look at that for us, please? >> >> Paul, thank you for the contribution. Would you please, next time, post a >> Jira so we are able to keep track of all contributions? >> >> Regards, >> Jo?o Mesquita >> FSComm Lead Developer >> >> >> On Fri, Jan 15, 2010 at 10:26 PM, Paul Li wrote: >> >>> The present FSComm VS2008 Project file need to be updated as attached >>> in order to build it with Visual Studio 2008 for the issues: >>> >>> (1) prefaccounts.h/prefaccounts.cpp need to be added to the project. >>> (2) "Generated Files" need to be filtered properly for Debug/Release >>> configuration, so that the proper files are used for the corresponding >>> configuration. >>> (3) Link to proper "FreeSwitchCore.lib", based on the selected >>> configuration. >>> (4) All intermediate files now go to $(IntDir), except for "ui_*.h". >>> >>> All these issues have been corrected with the attached .proj file. >>> >>> Thanks for sharing the app! >>> >>> _______________________________________________ >>> 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 >> >> > > _______________________________________________ > 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/20100121/e0be7c78/attachment-0001.html From plite2012 at gmail.com Wed Jan 20 22:07:00 2010 From: plite2012 at gmail.com (Paul Li) Date: Thu, 21 Jan 2010 00:07:00 -0600 Subject: [Freeswitch-dev] FSComm Custom Parameters and Crashes Message-ID: Custom Parameter tab is having another problem and crashes under the following situation. If a gateway, gw1, has two custom params, and the second gateway, gw2, has zero or one custom param. After you first try to edit gw1, and then cancel to close, then you proceed to edit gw2, you will see blank row(s). When click OK, FSComm crashes, which might resulted from the blank row(s) in the table. Jo?o, it appears that you did not completely clears the entries for the Advance (custom) tab. Other than this problem, everything else looks good, thanks for your hard work. Next time, I will try to log any issue with Jira. From ste.bossi at gmail.com Thu Jan 21 01:45:14 2010 From: ste.bossi at gmail.com (stefano bossi) Date: Thu, 21 Jan 2010 10:45:14 +0100 Subject: [Freeswitch-dev] thesis on live sip calls migration In-Reply-To: <8ab521c31001190604n600f68a7h6f895eb6ea88acf9@mail.gmail.com> References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com> <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> <8ab521c31001190604n600f68a7h6f895eb6ea88acf9@mail.gmail.com> Message-ID: <8ab521c31001210145t2205daceg4175bcb0b970d6d6@mail.gmail.com> Hi, after a deeper investigation on cseq I can can state that: - there are 2 cseq for each channel: in in my tests I substituted the variable name sip_cseq with sip_incoming_cseq when it's read from nua_i_* events and sip_outgoing_cseq when it's read from nua_r_* - in order to resurrect a call I need only the sip_outgoing_cseq, in the most part of cases this field remain unset for inbound channels ( so I can use a random one during migration) and it is equal to the invite one for outbound channel. - things go in different way when FS sends in-session requests to endpoints. In these cases I think it can be resolved by reading the sip_outgoing_cseq on nua_r_info, nua_r_prack and nua_r_update. Obviously the sip_incoming_cseq is increased on nua_i_info, nua_i_prack and nua_i_update but this field it's not useful for migrating a call. Maybe it will be necessary to fire some new switch custom events to give access to these updated field from event_socket. I'm here for any further explanations bye On Tue, Jan 19, 2010 at 3:04 PM, stefano bossi wrote: > Hi Anthony, > > your last commit on sofia.c adds a very helpful function to extract > sip_to_tag and sip_from_tag.. > I was working on something similar but I also read the Cseq.. you can do > this with these 3 lines of code > > if (sip->sip_cseq && sip->sip_cseq->cs_seq) { > const char *sip_cseq = switch_core_session_sprintf(session,"%d", > sip->sip_cseq->cs_seq); > switch_channel_set_variable(channel, "sip_cseq", sip_cseq); > } > > (It compiles, I hope it is correct too :) > > This cseq can be used for calls failover in the most part of cases, I only > wonder if this is still true after any other request during dialog than > INVITE. > > bye > > On Mon, Jan 4, 2010 at 5:16 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Hi, >> When would you like to continue this conversation? >> >> >> On Fri, Dec 18, 2009 at 4:38 AM, stefano bossi wrote: >> >>> Hi all, >>> >>> as promised in #freeswitch-dev I'll try to explain better what I've done >>> and I would like to achieve for my thesis. >>> >>> >>> At the moment I successfully patched sofsip_cli and now it can do >>> something near active call migration :) >>> Thanks to the versatility of nua api I could achieve my objective with >>> little effort. >>> Basically I call someone, then I kill sofsip_cli and I restart it. I >>> launch my "reinvite" command and the call can continue normally. The called >>> client doesn't realize what happened. >>> In this case I write call data in a file and when I launch the reinvite I >>> read directly from this file. >>> Sofsip_cli sends an INVITE message with some TAG appended to the >>> nua_handle. So the called phone thinks to receive an in-session invite and >>> simply refreshes the call, but Sofsip_cli instead allocates all the stuff >>> for a new call. In this way I can avoid to modify directly the RTP part. It >>> seems to be all simpler. >>> >>> I even wrote a little module for FS able to monitor(using the SIP OPTION >>> message) the life of a specific sofia profile.. It's just a proof of concept >>> but it can do failover(without live call migration) between 2 machines in >>> about 50ms. >>> This is possible with these actions: >>> >>> - registrations are shared with odbc >>> >>> >>> - on the start of the module(present only on the backup machine): >>> >>> >>> 1. I set an arp rule blocking the arp response for the virtual IP >>> (set on the primary machine) >>> 2. I set the virtual IP on the backup machine( but no one knows >>> thanks to the arp rule and so I can bind to vIP) >>> 3. I prepare and run the sip profile (on the backup)... loading here >>> the profile permits to save a lot of time during reaction >>> >>> >>> - on the reaction >>> >>> >>> 1. I remove the arp rule >>> 2. I send a gratuitous arp request >>> >>> As I said in about 50ms sip clients can call again. Maybe this time can >>> decrease using NETLINK socket for the arp table. >>> >>> The union of these 2 works will give us a very fast "live profile >>> migration" :D >>> >>> There some points to discuss: >>> >>> - switch_core_session_resurrect_channel: I didn't know this function, >>> I need to understand what it offers, maybe tomorrow ;) >>> - the propagation of call state variables: my first idea was to use >>> the multicast events adding some headers to send all the necessary data (but >>> anthm proposed XML) >>> - I thought only to sofia aspects.. >>> >>> In next days I'll try to understand where to hook in FS to try these >>> ideas. When I'll have something ready (and without very very big errors:) >>> I'll be happy to send you the patch. >>> This is my first work on something real like FS.. I'm opened to any kind >>> of suggestions!* >>> >>> *please contact me for further clarifications >>> >>> Thanks >>> >>> Stefano Bossi >>> >>> >>> >>> _______________________________________________ >>> 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100121/2c657237/attachment.html From vipkilla at gmail.com Thu Jan 21 05:52:17 2010 From: vipkilla at gmail.com (vip killa) Date: Thu, 21 Jan 2010 08:52:17 -0500 Subject: [Freeswitch-dev] fscomm with celt In-Reply-To: References: <957f61371001200655t5ccf1d8au3bcebf98a7f1368a@mail.gmail.com> <5800526b1001201016ubb2bd7r60caa1850277e6c9@mail.gmail.com> Message-ID: <957f61371001210552xe6d573mac4f4c80c40e0e9c@mail.gmail.com> yeah. nice work. thanks 2010/1/20 Jo?o Mesquita > Like usual, Carlos is tha man. > > Thanks Carlos! > > Regards, > Jo?o Mesquita > FSComm Developer > > > > On Wed, Jan 20, 2010 at 4:16 PM, Carlos Talbot wrote: > >> Until now that was not possible because there was no VS project file to >> build CELT on Windows >> >> I've just committed updates to resolve this. I've also recompiled FSComm >> with mod_celt and am uploading the most recent install file. >> >> On Wed, Jan 20, 2010 at 8:55 AM, vip killa wrote: >> >>> has anyone built fscomm with the celt codec for windows? >>> >>> _______________________________________________ >>> 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 >> >> > > _______________________________________________ > 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/20100121/5c292ea9/attachment.html From mardy at voysys.com Thu Jan 21 06:03:40 2010 From: mardy at voysys.com (Mardy Marshall) Date: Thu, 21 Jan 2010 09:03:40 -0500 Subject: [Freeswitch-dev] Problem with event_socket application when serving as a consultative transfer target In-Reply-To: <191c3a031001201611k7c8a5029u26e80f4eb1b75f2b@mail.gmail.com> References: <191c3a031001201611k7c8a5029u26e80f4eb1b75f2b@mail.gmail.com> Message-ID: <89FD5D65-AF79-47D0-BB05-DA870BDE883A@voysys.com> I have been trying to use filters in place of "myevents" and even with a basic outbound event_socket session I am unable to receive any events at all. Also, why is the "events" command only allowed for an inbound session? I would think that in my case, I would want to be issuing an "events plain all" command followed by UUID specific filters. On Jan 20, 2010, at 7:11 PM, Anthony Minessale wrote: > you could try using the filter command to filter on the various uuid rather than locking it on to 1 channel with myevents > essentially once it's transferred your leg is dead so you can stick around on the socket if you want but you can't control the other channel. > > If filter does not work we could look at a way to myevents over to another session but it may be tricky and involve a bounty. > > > On Wed, Jan 20, 2010 at 5:14 PM, Mardy Marshall wrote: > I'm having a problem with an event_socket application which is the transfer target of a consultative transfer. When the transferor begins the transfer process, the outbound event_socket session is created and everything looks normal. When the transfer is completed, switch_ivr_bridge fires a CHANNEL_EXECUTE / "uuid_bridge" event, supplying a new UUID for the "Other-Leg" which corresponds to the transferee. From this point on, I no longer receive events specific to this new UUID, such as DTMF events or CHANNEL_EXECUTE_COMPLETE events after issuing "playback" commands. Regarding the "playback" command, if I direct the command to the "Other-Leg" UUID, I get audio, but no events when it is done. On the other hand, if I direct the command to the original UUID, the one corresponding to the transferrer call leg, I get no audio, but do get completion events. I have tried re-issuing a "myevents" command using the new UUID, but the command fails due to it only being allowed to be called once per session. > > Am I missing something in order to handle the transfer or have I run into a bug? > > Thanks, > > -Mardy > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100121/98165ead/attachment-0001.html From mardy at voysys.com Thu Jan 21 07:54:00 2010 From: mardy at voysys.com (Mardy Marshall) Date: Thu, 21 Jan 2010 10:54:00 -0500 Subject: [Freeswitch-dev] Problem with event_socket application when serving as a consultative transfer target In-Reply-To: <89FD5D65-AF79-47D0-BB05-DA870BDE883A@voysys.com> References: <191c3a031001201611k7c8a5029u26e80f4eb1b75f2b@mail.gmail.com> <89FD5D65-AF79-47D0-BB05-DA870BDE883A@voysys.com> Message-ID: I went ahead and hacked mod_event_socket.c to enable the "event" command for both inbound and outbound sessions and now everything is working, including the consultative transfer target case. Should the "event" command really be restricted to only inbound sessions? -Mardy On Jan 21, 2010, at 9:03 AM, Mardy Marshall wrote: > I have been trying to use filters in place of "myevents" and even with a basic outbound event_socket session I am unable to receive any events at all. > > Also, why is the "events" command only allowed for an inbound session? I would think that in my case, I would want to be issuing an "events plain all" command followed by UUID specific filters. > > > On Jan 20, 2010, at 7:11 PM, Anthony Minessale wrote: > >> you could try using the filter command to filter on the various uuid rather than locking it on to 1 channel with myevents >> essentially once it's transferred your leg is dead so you can stick around on the socket if you want but you can't control the other channel. >> >> If filter does not work we could look at a way to myevents over to another session but it may be tricky and involve a bounty. >> >> >> On Wed, Jan 20, 2010 at 5:14 PM, Mardy Marshall wrote: >> I'm having a problem with an event_socket application which is the transfer target of a consultative transfer. When the transferor begins the transfer process, the outbound event_socket session is created and everything looks normal. When the transfer is completed, switch_ivr_bridge fires a CHANNEL_EXECUTE / "uuid_bridge" event, supplying a new UUID for the "Other-Leg" which corresponds to the transferee. From this point on, I no longer receive events specific to this new UUID, such as DTMF events or CHANNEL_EXECUTE_COMPLETE events after issuing "playback" commands. Regarding the "playback" command, if I direct the command to the "Other-Leg" UUID, I get audio, but no events when it is done. On the other hand, if I direct the command to the original UUID, the one corresponding to the transferrer call leg, I get no audio, but do get completion events. I have tried re-issuing a "myevents" command using the new UUID, but the command fails due to it only being allowed to be called once per session. >> >> Am I missing something in order to handle the transfer or have I run into a bug? >> >> Thanks, >> >> -Mardy >> >> > > _______________________________________________ > 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/20100121/40591f6b/attachment.html From plite2012 at gmail.com Thu Jan 21 22:30:57 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 22 Jan 2010 00:30:57 -0600 Subject: [Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan Message-ID: It is a very interesting and challenging issue! What I am trying to achieve is playback a message (notice) to PSTN telephones with Gizmo5 as service provider with the originate command as follows: $ originate sofia/gateway/gizmo/0019876543210 5001 (Case 1) $ originate sofia/gateway/gizmo/0019876543210 5002 (Case 2) I can hear the playback from extension 5001 in the first case, while I could not hear anything from extension 5002 in the second case. The difference between them lies in that extension 5002 transfer the call when the PSTN phone is connected. Please see the dial plan below. Both dial plans are basically adapted from examples found in the online documents. Both 5001 and 5002 work for my two other voip service providers. On the surface, it appears to be Gizmo5's problem. But, I am wondering if there is any issue on the FreeSwitch side. My main goal is to delay playback until the end user has picked up the phone. I have been looking for a way to do this but ended up with nothing so far. I have tried setting {ignore_early_media=true}, but that did not help either. I am welcome any suggestion on how to achieve this. In the meanwhile, I would like point out this issue. From mike at jerris.com Thu Jan 21 23:21:41 2010 From: mike at jerris.com (Michael Jerris) Date: Fri, 22 Jan 2010 02:21:41 -0500 Subject: [Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan In-Reply-To: References: Message-ID: <31D17703-E8D2-403A-A83C-93B8CB490AFD@jerris.com> I suggest you look at the logs and compare the working call to 5002 from one carrier with the non working with gizmo and see what is different. Mike On Jan 22, 2010, at 1:30 AM, Paul Li wrote: > It is a very interesting and challenging issue! What I am trying to > achieve is playback a message (notice) to PSTN telephones with Gizmo5 > as service provider with the originate command as follows: > > $ originate sofia/gateway/gizmo/0019876543210 5001 (Case 1) > > $ originate sofia/gateway/gizmo/0019876543210 5002 (Case 2) > > I can hear the playback from extension 5001 in the first case, while I > could not hear anything from extension 5002 in the second case. The > difference between them lies in that extension 5002 transfer the call > when the PSTN phone is connected. Please see the dial plan below. Both > dial plans are basically adapted from examples found in the online > documents. Both 5001 and 5002 work for my two other voip service > providers. On the surface, it appears to be Gizmo5's problem. But, I > am wondering if there is any issue on the FreeSwitch side. From mlistsdm at gmail.com Thu Jan 21 13:27:29 2010 From: mlistsdm at gmail.com (Dmitry M) Date: Thu, 21 Jan 2010 23:27:29 +0200 Subject: [Freeswitch-dev] FSComm preferences Message-ID: <4B58C6C1.20603@gmail.com> Checking phones preferences I noticed Caller id Name and Number located in General Preferences. IMO it'll be more natural to move them into Account. I expect account may have different proto (SIP,IAX,H323,...), different caller id, different codec settings, etc. Dmitry From msc at freeswitch.org Fri Jan 22 08:29:46 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Jan 2010 08:29:46 -0800 Subject: [Freeswitch-dev] FreeSWITCH Weekly Call Starting in 30 min Message-ID: <87f2f3b91001220829k32f60609kbd76c07c93db435e@mail.gmail.com> Hello all, The weekly conference call will be starting in a bit. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22 I'll be on just after 9am my time as I have to get my kids off to school. Talk to you all soon. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100122/f3d3bb67/attachment.html From brian at freeswitch.org Fri Jan 22 09:54:22 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 22 Jan 2010 11:54:22 -0600 Subject: [Freeswitch-dev] IAX2 Support Removed. Message-ID: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org> Due to lack of support for the libiax2 being updated to support the newer protocol changes and the lack of interest from anyone willing to actually work on it. I have moved mod_iax to unsupported where it will stay until someone steps up to rewrite a new IAX2 lib. Thanks, Brian From plite2012 at gmail.com Fri Jan 22 10:23:27 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 22 Jan 2010 12:23:27 -0600 Subject: [Freeswitch-dev] How to delay playback during an originate call? Message-ID: I really need some help/advice on how to delay playback during the following originate calls: $ originate {ignore_early_media=true}sofia/gateway/name/number &playback(message.wav) $ originate sofia/example/1000 at somewhere.com &javascript(test.js) I am hoping playback of message.wav or application (calling test.js) is invoked only after the call is established (connected). Is it possible to add some flag or channel variable, like media_bug_answer_req, to delay playback or call the JavaScript, Lua script, etc.? From plite2012 at gmail.com Fri Jan 22 23:10:58 2010 From: plite2012 at gmail.com (Paul Li) Date: Sat, 23 Jan 2010 01:10:58 -0600 Subject: [Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan In-Reply-To: References: Message-ID: By comparing the log from a working service provider (Working) to that from Gizmo5, I have found out the problem might be Gizmo5 (service provider). While FreeSWITCH was performing TRANSFER, Gizmo5: Channel sofia/softphone/001x...x entering state [completing][200] Working: Channel sofia/softphone/001x...x entering state [proceeding][180] Gizmo5: Channel sofia/softphone/001x...x entering state [ready][200] Working: Channel sofia/softphone/001x...x entering state [proceeding][180] Gizmo5: [Channel sofia/softphone/001x...x] has been answered Working: Pre-Answer sofia/softphone/001x...x Both began "Running State Change CS_ROUTING". After routing was successful, the working service provider finally enter [200] state. Working: Channel sofia/softphone/001x...x entering state [completing][200] Working: Channel sofia/softphone/001x...x entering state [ready][200] At this point, the working service provider finally connects sofia/softphone/001x...x to the playback channel. I am wondering what response Gizmo5 sent to FreeSWITCH and led FreeSWITCH to think the call prematurally entered state [completing][200] and [ready][200]. On Fri, Jan 22, 2010 at 12:30 AM, Paul Li wrote: > It is a very interesting and challenging issue! What I am trying to > achieve is playback a message (notice) to PSTN telephones with Gizmo5 > as service provider with the originate command as follows: > > $ originate sofia/gateway/gizmo/0019876543210 5001 ? ? (Case 1) > > $ originate sofia/gateway/gizmo/0019876543210 5002 ? ? (Case 2) > > I can hear the playback from extension 5001 in the first case, while I > could not hear anything from extension 5002 in the second case. The > difference between them lies in that extension 5002 transfer the call > when the PSTN phone is connected. Please see the dial plan below. Both > dial plans are basically adapted from examples found in the online > documents. Both 5001 and 5002 work for my two other voip service > providers. On the surface, it appears to be Gizmo5's problem. But, I > am wondering if there is any issue on the FreeSwitch side. > > My main goal is to delay playback until the end user has picked up the > phone. I have been looking for a way to do this but ended up with > nothing so far. I have tried setting {ignore_early_media=true}, but > that did not help either. > > I am welcome any suggestion on how to achieve this. In the meanwhile, > I would like point out this issue. > > ? ? ? ? > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > ? ? ? ? > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > ? ? ? ? > ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > From dujinfang at gmail.com Sat Jan 23 04:58:13 2010 From: dujinfang at gmail.com (Seven Du) Date: Sat, 23 Jan 2010 20:58:13 +0800 Subject: [Freeswitch-dev] How to use FSComm to make a call? In-Reply-To: References: <23f91031001181812h2285c520vcbd73b6e18625fa8@mail.gmail.com> Message-ID: <23f91031001230458i636e0967v6281df4bb3de180c@mail.gmail.com> > FSComm is getting big so some organization is needed. > Regards, > Jo?o Mesquita > FSComm Developer Yeah, agree. I will let you know before I work on a new feature. From jmesquita at freeswitch.org Sat Jan 23 08:19:36 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 23 Jan 2010 14:19:36 -0200 Subject: [Freeswitch-dev] FSComm preferences In-Reply-To: <4B58C6C1.20603@gmail.com> References: <4B58C6C1.20603@gmail.com> Message-ID: I fully agree with you and this is the way it is supposed to be coded, but we are still under heavy development and I just didn't have time to make it work properly. I will start coding that right now. Regards, Jo?o Mesquita FSComm Developer On Thu, Jan 21, 2010 at 7:27 PM, Dmitry M wrote: > Checking phones preferences I noticed > Caller id Name and Number located in General Preferences. > > IMO it'll be more natural to move them into Account. > I expect account may have different proto (SIP,IAX,H323,...), > different caller id, different codec settings, etc. > > Dmitry > > > _______________________________________________ > 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/20100123/1f1769e3/attachment.html From jmesquita at freeswitch.org Sat Jan 23 08:24:39 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 23 Jan 2010 14:24:39 -0200 Subject: [Freeswitch-dev] Call for help on FSComm Message-ID: FreeSWITCH?ers, I believe that everyone already heard about FSComm by now. Development is moving fast and features are being added very fast. Nonetheless, we need help to get this software the way we want. If you are a GUI Designer or have any design skills and want to contribute, please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if you like. I believe everyone wants a sexy looking softphone with all the features FreeSWITCH? is able to provide. Regards, Jo?o Mesquita FSComm Developer GTalk: jmesquita at gmail.com PayPal: jmesquita at gmail.com IRC: jmesquita on #fscomm @freenode -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100123/78abcde2/attachment.html From jmesquita at freeswitch.org Sat Jan 23 11:14:21 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 23 Jan 2010 17:14:21 -0200 Subject: [Freeswitch-dev] FSComm Custom Parameters and Crashes In-Reply-To: References: Message-ID: Paul, do you mind posting a Jira with the proper log information? Appreciated. Regards, Jo?o Mesquita FSComm Developer On Thu, Jan 21, 2010 at 4:07 AM, Paul Li wrote: > Custom Parameter tab is having another problem and crashes under the > following situation. If a gateway, gw1, has two custom params, and the > second gateway, gw2, has zero or one custom param. After you first try > to edit gw1, and then cancel to close, then you proceed to edit gw2, > you will see blank row(s). When click OK, FSComm crashes, which might > resulted from the blank row(s) in the table. > > Jo?o, it appears that you did not completely clears the entries for > the Advance (custom) tab. Other than this problem, everything else > looks good, thanks for your hard work. Next time, I will try to log > any issue with Jira. > > _______________________________________________ > 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/20100123/c67a2e42/attachment.html From mike at jerris.com Sat Jan 23 11:18:39 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 23 Jan 2010 14:18:39 -0500 Subject: [Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan In-Reply-To: References: Message-ID: <6D78CDA3-E56C-42A7-9BAF-4FC34B45A792@jerris.com> That means they answered the call already. Is that accurate? On Jan 23, 2010, at 2:10 AM, Paul Li wrote: > By comparing the log from a working service provider (Working) to that > from Gizmo5, I have found out the problem might be Gizmo5 (service > provider). While FreeSWITCH was performing TRANSFER, > > Gizmo5: Channel sofia/softphone/001x...x entering state [completing][200] > Working: Channel sofia/softphone/001x...x entering state [proceeding][180] > > Gizmo5: Channel sofia/softphone/001x...x entering state [ready][200] > Working: Channel sofia/softphone/001x...x entering state [proceeding][180] > > Gizmo5: [Channel sofia/softphone/001x...x] has been answered > Working: Pre-Answer sofia/softphone/001x...x > > Both began "Running State Change CS_ROUTING". After routing was > successful, the working service provider finally enter [200] state. > > Working: Channel sofia/softphone/001x...x entering state [completing][200] > Working: Channel sofia/softphone/001x...x entering state [ready][200] > > At this point, the working service provider finally connects > sofia/softphone/001x...x to the playback channel. I am wondering what > response Gizmo5 sent to FreeSWITCH and led FreeSWITCH to think the > call prematurally entered state [completing][200] and [ready][200]. > > > On Fri, Jan 22, 2010 at 12:30 AM, Paul Li wrote: >> It is a very interesting and challenging issue! What I am trying to >> achieve is playback a message (notice) to PSTN telephones with Gizmo5 >> as service provider with the originate command as follows: >> >> $ originate sofia/gateway/gizmo/0019876543210 5001 (Case 1) >> >> $ originate sofia/gateway/gizmo/0019876543210 5002 (Case 2) >> >> I can hear the playback from extension 5001 in the first case, while I >> could not hear anything from extension 5002 in the second case. The >> difference between them lies in that extension 5002 transfer the call >> when the PSTN phone is connected. Please see the dial plan below. Both >> dial plans are basically adapted from examples found in the online >> documents. Both 5001 and 5002 work for my two other voip service >> providers. On the surface, it appears to be Gizmo5's problem. But, I >> am wondering if there is any issue on the FreeSwitch side. >> >> My main goal is to delay playback until the end user has picked up the >> phone. I have been looking for a way to do this but ended up with >> nothing so far. I have tried setting {ignore_early_media=true}, but >> that did not help either. >> >> I am welcome any suggestion on how to achieve this. In the meanwhile, >> I would like point out this issue. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > > _______________________________________________ > 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 ste.bossi at gmail.com Sat Jan 23 11:43:21 2010 From: ste.bossi at gmail.com (Stefano Bossi) Date: Sat, 23 Jan 2010 20:43:21 +0100 Subject: [Freeswitch-dev] thesis on live sip calls migration In-Reply-To: <8ab521c31001190604n600f68a7h6f895eb6ea88acf9@mail.gmail.com> References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com> <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com> <191c3a031001040816v4eab4f5nf86ee3fe96b1ef93@mail.gmail.com> <8ab521c31001190604n600f68a7h6f895eb6ea88acf9@mail.gmail.com> Message-ID: <1264275801.4615.129.camel@husky-desktop> Hi, I have a little patch to enable sofia_glue_do_invite to send a (fake)reinvite instead of a regular invite. The changes allow to set to_tag,from_tag and cseq from the originate command. To simply try this function I do these steps: 1 start freeswitch 2 sofia profile internal siptrace on 3 originate sofia/internal/1000%mydomain.it &park() 4 when the call is active I read from the 200 OK header the cseq, the from_tag, the to_tag and the call-id 5 create the command for resurrecting the call on a text editor: originate originate {sip_outgoing_call_id=798e29c9-82f7-122d-2b9a-00163e2b53e5,sip_invite_from_tag=UgDZZegc8Q2rF,sip_invite_to_tag=947294406,sip_invite_cseq=126018935}sofia/internal/1000%mydomain.it &park() (those are my last values, put yours) 6 at this point, with the call still active, kill -9 freeswitch 7 start freeswitch immediately 8 launch the command previously prepared 9 the phone still thinks to be connected with the old FS instance 10 now you can hangup and, if you enable the siptrace again, see the correct tear down of the call. Obviously this can be done with 2 call legs but it is a little more complicated. If someone is interested I can explain how to do. And of course if you have another identical machine with the same IP, with registrations sharing enabled, ready to wake up after the crash of the first one you can do the same procedure :) I hope the procedure is clear and sorry for my poor English bye Stefano On mar, 2010-01-19 at 15:04 +0100, stefano bossi wrote: > Hi Anthony, > > > your last commit on sofia.c adds a very helpful function to extract > sip_to_tag and sip_from_tag.. > I was working on something similar but I also read the Cseq.. you can > do this with these 3 lines of code > > > if (sip->sip_cseq && sip->sip_cseq->cs_seq) { > const char *sip_cseq = switch_core_session_sprintf(session,"%d", > sip->sip_cseq->cs_seq); > switch_channel_set_variable(channel, "sip_cseq", sip_cseq); > } > > > (It compiles, I hope it is correct too :) > > > This cseq can be used for calls failover in the most part of cases, I > only wonder if this is still true after any other request during > dialog than INVITE. > > > bye > > On Mon, Jan 4, 2010 at 5:16 PM, Anthony Minessale > wrote: > Hi, > When would you like to continue this conversation? > > > > On Fri, Dec 18, 2009 at 4:38 AM, stefano bossi > wrote: > > > Hi all, > > as promised in #freeswitch-dev I'll try to explain > better what I've done and I would like to achieve for > my thesis. > > > At the moment I successfully patched sofsip_cli and > now it can do something near active call migration :) > Thanks to the versatility of nua api I could achieve > my objective with little effort. > Basically I call someone, then I kill sofsip_cli and I > restart it. I launch my "reinvite" command and the > call can continue normally. The called client doesn't > realize what happened. > In this case I write call data in a file and when I > launch the reinvite I read directly from this file. > Sofsip_cli sends an INVITE message with some TAG > appended to the nua_handle. So the called phone thinks > to receive an in-session invite and simply refreshes > the call, but Sofsip_cli instead allocates all the > stuff for a new call. In this way I can avoid to > modify directly the RTP part. It seems to be all > simpler. > > I even wrote a little module for FS able to > monitor(using the SIP OPTION message) the life of a > specific sofia profile.. It's just a proof of concept > but it can do failover(without live call migration) > between 2 machines in about 50ms. > This is possible with these actions: > * registrations are shared with odbc > * on the start of the module(present only on the > backup machine): > 1. I set an arp rule blocking the arp response > for the virtual IP (set on the primary > machine) > 2. I set the virtual IP on the backup > machine( but no one knows thanks to the arp > rule and so I can bind to vIP) > 3. I prepare and run the sip profile (on the > backup)... loading here the profile permits to > save a lot of time during reaction > * on the reaction > 1. I remove the arp rule > 2. I send a gratuitous arp request > As I said in about 50ms sip clients can call again. > Maybe this time can decrease using NETLINK socket for > the arp table. > > The union of these 2 works will give us a very fast > "live profile migration" :D > > > There some points to discuss: > * switch_core_session_resurrect_channel: I > didn't know this function, I need to > understand what it offers, maybe tomorrow ;) > * the propagation of call state variables: my > first idea was to use the multicast events > adding some headers to send all the necessary > data (but anthm proposed XML) > * I thought only to sofia aspects.. > In next days I'll try to understand where to hook in > FS to try these ideas. When I'll have something ready > (and without very very big errors:) I'll be happy to > send you the patch. > This is my first work on something real like FS.. I'm > opened to any kind of suggestions! > > please contact me for further clarifications > > Thanks > > Stefano Bossi > > > > > > > _______________________________________________ > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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 -------------- next part -------------- Index: src/mod/endpoints/mod_sofia/sofia_glue.c =================================================================== --- src/mod/endpoints/mod_sofia/sofia_glue.c (revision 16485) +++ src/mod/endpoints/mod_sofia/sofia_glue.c (working copy) @@ -1457,6 +1457,10 @@ sofia_destination_t *dst = NULL; sofia_cid_type_t cid_type = tech_pvt->profile->cid_type; + //int failover_flag = 0; + sip_cseq_t *cseq = NULL; + //char *sip_invite_cseq = NULL; + rep = switch_channel_get_variable(channel, SOFIA_REPLACES_HEADER); switch_assert(tech_pvt != NULL); @@ -1503,8 +1507,11 @@ const char *invite_from_params = switch_channel_get_variable(tech_pvt->channel, "sip_invite_from_params"); const char *from_var = switch_channel_get_variable(tech_pvt->channel, "sip_from_uri"); const char *from_display = switch_channel_get_variable(tech_pvt->channel, "sip_from_display"); - - + const char *invite_from_tag = switch_channel_get_variable(tech_pvt->channel, "sip_invite_from_tag"); + const char *invite_to_tag = switch_channel_get_variable(tech_pvt->channel, "sip_invite_to_tag"); + sip_from_t *from = NULL; + sip_to_t *to = NULL; + if (zstr(tech_pvt->dest)) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_ERROR, "URL Error!\n"); return SWITCH_STATUS_FALSE; @@ -1544,6 +1551,8 @@ sipip); } + + if (from_var) { if (strncasecmp(from_var, "sip:", 4) || strncasecmp(from_var, "sips:", 5)) { use_from_str = switch_core_session_strdup(tech_pvt->session, from_var); @@ -1669,11 +1678,19 @@ } } + from = sip_from_make(tech_pvt->profile->home, from_str); + if (!zstr(invite_from_tag)) + sip_from_tag(tech_pvt->profile->home,from,invite_from_tag); + + to = sip_to_make(tech_pvt->profile->home, to_str); + if (!zstr(invite_to_tag)) + sip_to_tag(tech_pvt->profile->home,to,invite_to_tag); + tech_pvt->nh = nua_handle(tech_pvt->profile->nua, NULL, NUTAG_URL(url_str), TAG_IF(call_id, SIPTAG_CALL_ID_STR(call_id)), - SIPTAG_TO_STR(to_str), - SIPTAG_FROM_STR(from_str), + SIPTAG_TO(to), + SIPTAG_FROM(from), SIPTAG_CONTACT_STR(invite_contact), TAG_END()); @@ -1825,7 +1842,12 @@ switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_DEBUG, "%s Setting proxy route to %s\n", route_uri, switch_channel_get_name(channel)); tech_pvt->route_uri = switch_core_session_strdup(tech_pvt->session, route_uri); } - + + if (( val = switch_channel_get_variable(tech_pvt->channel, "sip_invite_cseq"))) { + uint32_t callsequence = (uint32_t)strtoul(val,NULL,10); + cseq = sip_cseq_create(tech_pvt->profile->home, callsequence, SIP_METHOD_INVITE); + } + switch_channel_clear_flag(channel, CF_MEDIA_ACK); if (sofia_use_soa(tech_pvt)) { @@ -1845,6 +1867,7 @@ TAG_IF(!zstr(route_uri), NUTAG_PROXY(route_uri)), TAG_IF(!zstr(route), SIPTAG_ROUTE_STR(route)), TAG_IF(tech_pvt->profile->minimum_session_expires, NUTAG_MIN_SE(tech_pvt->profile->minimum_session_expires)), + TAG_IF(cseq,SIPTAG_CSEQ(cseq)), SOATAG_ADDRESS(tech_pvt->adv_sdp_audio_ip), SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str), SOATAG_REUSE_REJECTED(1), @@ -1871,6 +1894,7 @@ TAG_IF(!zstr(route_uri), NUTAG_PROXY(route_uri)), TAG_IF(!zstr(route), SIPTAG_ROUTE_STR(route)), TAG_IF(tech_pvt->profile->minimum_session_expires, NUTAG_MIN_SE(tech_pvt->profile->minimum_session_expires)), + TAG_IF(cseq,SIPTAG_CSEQ(cseq)), NUTAG_MEDIA_ENABLE(0), SIPTAG_CONTENT_TYPE_STR("application/sdp"), SIPTAG_PAYLOAD_STR(tech_pvt->local_sdp_str), From plite2012 at gmail.com Sat Jan 23 16:41:53 2010 From: plite2012 at gmail.com (Paul Li) Date: Sat, 23 Jan 2010 18:41:53 -0600 Subject: [Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan In-Reply-To: References: Message-ID: I believe so. I have logged a ticket with Gizmo5. I hope they could resolve this issue soon. On Sat Jan 23 11:18:39 PST 2010, Michael Jerris wrote: > That means they answered the call already. Is that accurate? On Sat, Jan 23, 2010 at 1:10 AM, Paul Li wrote: > By comparing the log from a working service provider (Working) to that > from Gizmo5, I have found out the problem might be Gizmo5 (service > provider). While FreeSWITCH was performing TRANSFER, > > Gizmo5: Channel sofia/softphone/001x...x entering state [completing][200] > Working: Channel sofia/softphone/001x...x entering state [proceeding][180] > > Gizmo5: Channel sofia/softphone/001x...x entering state [ready][200] > Working: Channel sofia/softphone/001x...x entering state [proceeding][180] > > Gizmo5: [Channel sofia/softphone/001x...x] has been answered > Working: Pre-Answer sofia/softphone/001x...x > > Both began "Running State Change CS_ROUTING". After routing was > successful, the working service provider finally enter [200] state. > > Working: Channel sofia/softphone/001x...x entering state [completing][200] > Working: Channel sofia/softphone/001x...x entering state [ready][200] > > At this point, the working service provider finally connects > sofia/softphone/001x...x to the playback channel. I am wondering what > response Gizmo5 sent to FreeSWITCH ?and led FreeSWITCH ?to think the > call prematurally entered state [completing][200] and [ready][200]. From vipkilla at gmail.com Mon Jan 25 14:55:47 2010 From: vipkilla at gmail.com (vip killa) Date: Mon, 25 Jan 2010 17:55:47 -0500 Subject: [Freeswitch-dev] fscomm Call failed with reason NO_ROUTE_DESTINATION. Message-ID: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> Hi, I've downloaded and installed the latest FSComm for windows. I am able to register to the freeswitch server but i cannot make a call. When i try an extension i know works (because i use it with ekiga) i get "Call with (####) failed with reason NO_ROUTE_DESTINATION". I've tried calling ####@server.ip. But nothing works. I can receive calls to the FSComm application. I'm running FSComm on windows xp 64. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100125/1723c75f/attachment.html From brian at freeswitch.org Mon Jan 25 15:10:47 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 25 Jan 2010 17:10:47 -0600 Subject: [Freeswitch-dev] fscomm Call failed with reason NO_ROUTE_DESTINATION. In-Reply-To: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> Message-ID: <28921A15-A185-450B-8871-11F6E8BD2113@freeswitch.org> try sip:number at server /b On Jan 25, 2010, at 4:55 PM, vip killa wrote: > Hi, I've downloaded and installed the latest FSComm for windows. I am able to register to the freeswitch server but i cannot make a call. When i try an extension i know works (because i use it with ekiga) i get "Call with (####) failed with reason NO_ROUTE_DESTINATION". I've tried calling ####@server.ip. But nothing works. I can receive calls to the FSComm application. > I'm running FSComm on windows xp 64. From jmesquita at freeswitch.org Mon Jan 25 16:50:43 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 25 Jan 2010 18:50:43 -0600 Subject: [Freeswitch-dev] fscomm Call failed with reason NO_ROUTE_DESTINATION. In-Reply-To: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> Message-ID: Get me logs if that does not work. Regards, Jo?o Mesquita FSComm Developer On Mon, Jan 25, 2010 at 4:55 PM, vip killa wrote: > Hi, I've downloaded and installed the latest FSComm for windows. I am able > to register to the freeswitch server but i cannot make a call. When i try an > extension i know works (because i use it with ekiga) i get "Call with (####) > failed with reason NO_ROUTE_DESTINATION". I've tried calling ####@server.ip. > But nothing works. I can receive calls to the FSComm application. > I'm running FSComm on windows xp 64. > > _______________________________________________ > 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/20100125/2288a1af/attachment.html From tonhudung at gmail.com Tue Jan 26 03:53:00 2010 From: tonhudung at gmail.com (Alex To) Date: Tue, 26 Jan 2010 19:53:00 +0800 Subject: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c In-Reply-To: References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> Message-ID: <001501ca9e7e$1e4ea960$5aebfc20$@com> Hi, Would somebody enlighten me on this trunk of code in the mentioned file at line 2273: if (local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } if (vdata) { char *var_array[1024] = { 0 }; int var_count = 0; if ((var_count = switch_separate_string(vdata, '|', var_array, (sizeof(var_array) / sizeof(var_array[0]))))) { int x = 0; for (x = 0; x < var_count; x++) { char *inner_var_array[2] = { 0 }; int inner_var_count; if ((inner_var_count = switch_separate_string(var_array[x], '=', inner_var_array, (sizeof(inner_var_array) / sizeof(inner_var_array[0])))) == 2) { switch_channel_set_variable(originate_status[i].peer_channel, inner_var_array[0], inner_var_array[1]); } } } } if (!local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } The code block inside if (local_clobber) and if (!local_clobber) does exactly the same thing. May I know is it by design and how does it actually work ? I am not sure what does it mean by localClobber so if someone can spare sometime explaining, I would really appreciate it. Thank you Best Regards Alex To -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100126/bc7b4753/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 26 07:27:11 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Jan 2010 09:27:11 -0600 Subject: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c In-Reply-To: <001501ca9e7e$1e4ea960$5aebfc20$@com> References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> <001501ca9e7e$1e4ea960$5aebfc20$@com> Message-ID: <191c3a031001260727h799b0bc3y1e3a328cc83c648@mail.gmail.com> local_clobber true|false (default false) whether or not to allow (per-leg) local variables inside [] to take precedence over originate global variables in {} {foo=value1,local_clobber=false}sofia/internal/100 at test.com ,[foo=value2]sofia/internal/101 at test.com foo will be value1 on both legs {foo=value1,local_clobber=true}sofia/internal/100 at test.com ,[foo=value2]sofia/internal/101 at test.com foo will be value1 on all legs except the 101 who set it to value2 On Tue, Jan 26, 2010 at 5:53 AM, Alex To wrote: > Hi, > > > > Would somebody enlighten me on this trunk of code in the mentioned file at > line 2273: > > > > if (local_clobber) { > > if (var_event) { > > switch_event_t *event; > > switch_event_header_t *header; > > /* install the vars from the {} params > */ > > for (header = var_event->headers; > header; header = header->next) { > > > switch_channel_set_variable(originate_status[i].peer_channel, header->name, > header->value); > > } > > switch_event_create(&event, > SWITCH_EVENT_CHANNEL_ORIGINATE); > > switch_assert(event); > > > switch_channel_event_set_data(originate_status[i].peer_channel, event); > > switch_event_fire(&event); > > } > > } > > > > if (vdata) { > > char *var_array[1024] = { 0 }; > > int var_count = 0; > > if ((var_count = > switch_separate_string(vdata, '|', var_array, (sizeof(var_array) / sizeof(var_array[0]))))) > { > > int x = 0; > > for (x = 0; x < var_count; x++) { > > char *inner_var_array[2] = { 0 > }; > > int inner_var_count; > > if ((inner_var_count = > > > switch_separate_string(var_array[x], '=', > > > inner_var_array, (sizeof(inner_var_array) / sizeof(inner_var_array[0])))) > == 2) { > > > > > switch_channel_set_variable(originate_status[i].peer_channel, > inner_var_array[0], inner_var_array[1]); > > } > > } > > } > > } > > > > if (!local_clobber) { > > if (var_event) { > > switch_event_t *event; > > switch_event_header_t *header; > > /* install the vars from the {} params > */ > > for (header = var_event->headers; > header; header = header->next) { > > > switch_channel_set_variable(originate_status[i].peer_channel, header->name, > header->value); > > } > > switch_event_create(&event, > SWITCH_EVENT_CHANNEL_ORIGINATE); > > switch_assert(event); > > > switch_channel_event_set_data(originate_status[i].peer_channel, event); > > switch_event_fire(&event); > > } > > } > > > > The code block inside if (local_clobber) and if (!local_clobber) does > exactly the same thing. May I know is it by design and how does it actually > work ? I am not sure what does it mean by localClobber so if someone can > spare sometime explaining, I would really appreciate it. > > > > Thank you > > > > Best Regards > > > > Alex To > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100126/48615382/attachment.html From laureen at laureen.at Tue Jan 26 08:07:34 2010 From: laureen at laureen.at (Roland Zagler) Date: Tue, 26 Jan 2010 17:07:34 +0100 Subject: [Freeswitch-dev] Estimated release date for 1.0.5 final Message-ID: <001101ca9ea1$ac52ef00$04f8cd00$@at> Hi everyone, does anyone already have an idea on when 1.0.5 final will be released? Or at least a timeframe... Thanks, Roland Zagler From mbrancaleoni at voismart.it Tue Jan 26 08:24:59 2010 From: mbrancaleoni at voismart.it (Matteo) Date: Tue, 26 Jan 2010 17:24:59 +0100 (CET) Subject: [Freeswitch-dev] Estimated release date for 1.0.5 final In-Reply-To: <001101ca9ea1$ac52ef00$04f8cd00$@at> Message-ID: <1878596023.10088.1264523099050.JavaMail.root@mx.voismart.com> Hi, if you need a stable release, you do not know freeswitch nor trust it ;) bye, mat From freeswitch at cartissolutions.com Tue Jan 26 14:26:16 2010 From: freeswitch at cartissolutions.com (Yossi Neiman) Date: Tue, 26 Jan 2010 16:26:16 -0600 Subject: [Freeswitch-dev] Estimated release date for 1.0.5 final In-Reply-To: <001101ca9ea1$ac52ef00$04f8cd00$@at> References: <001101ca9ea1$ac52ef00$04f8cd00$@at> Message-ID: <4B5F6C08.1090207@cartissolutions.com> The standard response is always "whenever it's ready". -- Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com On 1/26/2010 10:07, Roland Zagler wrote: > Hi everyone, > > does anyone already have an idea on when 1.0.5 final will be released? > Or at least a timeframe... > > Thanks, Roland Zagler > > > > _______________________________________________ > 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 msc at freeswitch.org Tue Jan 26 14:46:50 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Jan 2010 14:46:50 -0800 Subject: [Freeswitch-dev] Estimated release date for 1.0.5 final In-Reply-To: <4B5F6C08.1090207@cartissolutions.com> References: <001101ca9ea1$ac52ef00$04f8cd00$@at> <4B5F6C08.1090207@cartissolutions.com> Message-ID: <87f2f3b91001261446l67285dd3td8ccf4ed05672c03@mail.gmail.com> On Tue, Jan 26, 2010 at 2:26 PM, Yossi Neiman < freeswitch at cartissolutions.com> wrote: > The standard response is always "whenever it's ready". > > I heard a rumor that it will be "soon" :) In the meantime the files at latest.freeswitch.org are definitely better than 1.0.4. Every day now is a new RC that is better than yesterday's. Definitely use the latest. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100126/8860e592/attachment.html From msc at freeswitch.org Tue Jan 26 15:54:22 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Jan 2010 15:54:22 -0800 Subject: [Freeswitch-dev] FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team Message-ID: <87f2f3b91001261554m2bfbbc45q21fc41c7d0a43b8@mail.gmail.com> Hello all, The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: http://www.freeswitch.org/node/234 Let's all show the team how much we appreciate them by giving them a well-deserved dinner! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100126/0b80b208/attachment.html From tonhudung at gmail.com Tue Jan 26 22:56:23 2010 From: tonhudung at gmail.com (Alex To) Date: Wed, 27 Jan 2010 14:56:23 +0800 Subject: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c In-Reply-To: <191c3a031001260727h799b0bc3y1e3a328cc83c648@mail.gmail.com> References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> <001501ca9e7e$1e4ea960$5aebfc20$@com> <191c3a031001260727h799b0bc3y1e3a328cc83c648@mail.gmail.com> Message-ID: <000c01ca9f1d$e749c580$b5dd5080$@com> Oh thank you Mike. I overlook the if (vdata) code block in between if(local_clobber) and if (!local_clobber). So actually if (!local_clobber) it will override the local variables by setting the global ones inside the {.}. So it's clear now. Thanks again. Alex To From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, January 26, 2010 11:27 PM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c local_clobber true|false (default false) whether or not to allow (per-leg) local variables inside [] to take precedence over originate global variables in {} {foo=value1,local_clobber=false}sofia/internal/100 at test.com,[foo=value2]sofi a/internal/101 at test.com foo will be value1 on both legs {foo=value1,local_clobber=true}sofia/internal/100 at test.com,[foo=value2]sofia /internal/101 at test.com foo will be value1 on all legs except the 101 who set it to value2 On Tue, Jan 26, 2010 at 5:53 AM, Alex To wrote: Hi, Would somebody enlighten me on this trunk of code in the mentioned file at line 2273: if (local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } if (vdata) { char *var_array[1024] = { 0 }; int var_count = 0; if ((var_count = switch_separate_string(vdata, '|', var_array, (sizeof(var_array) / sizeof(var_array[0]))))) { int x = 0; for (x = 0; x < var_count; x++) { char *inner_var_array[2] = { 0 }; int inner_var_count; if ((inner_var_count = switch_separate_string(var_array[x], '=', inner_var_array, (sizeof(inner_var_array) / sizeof(inner_var_array[0])))) == 2) { switch_channel_set_variable(originate_status[i].peer_channel, inner_var_array[0], inner_var_array[1]); } } } } if (!local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } The code block inside if (local_clobber) and if (!local_clobber) does exactly the same thing. May I know is it by design and how does it actually work ? I am not sure what does it mean by localClobber so if someone can spare sometime explaining, I would really appreciate it. Thank you Best Regards Alex To _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100127/df24c3bd/attachment-0001.html From tonhudung at gmail.com Wed Jan 27 00:41:15 2010 From: tonhudung at gmail.com (Alex To) Date: Wed, 27 Jan 2010 16:41:15 +0800 Subject: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c In-Reply-To: <191c3a031001260727h799b0bc3y1e3a328cc83c648@mail.gmail.com> References: <957f61371001251455g17df0c10p20d59f5612236951@mail.gmail.com> <001501ca9e7e$1e4ea960$5aebfc20$@com> <191c3a031001260727h799b0bc3y1e3a328cc83c648@mail.gmail.com> Message-ID: <001301ca9f2c$82fee240$88fca6c0$@com> Ignore my previous message. Wanted to say thanks Anthony but I wrote "Mike" . I think I wrote it in a hurry. Sorry for the mistake and thanks Anthony for your help. :D Regards Alex To From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, January 26, 2010 11:27 PM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Need help on switch_ivr_originate(...) in switch_ivr_originate.c local_clobber true|false (default false) whether or not to allow (per-leg) local variables inside [] to take precedence over originate global variables in {} {foo=value1,local_clobber=false}sofia/internal/100 at test.com,[foo=value2]sofi a/internal/101 at test.com foo will be value1 on both legs {foo=value1,local_clobber=true}sofia/internal/100 at test.com,[foo=value2]sofia /internal/101 at test.com foo will be value1 on all legs except the 101 who set it to value2 On Tue, Jan 26, 2010 at 5:53 AM, Alex To wrote: Hi, Would somebody enlighten me on this trunk of code in the mentioned file at line 2273: if (local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } if (vdata) { char *var_array[1024] = { 0 }; int var_count = 0; if ((var_count = switch_separate_string(vdata, '|', var_array, (sizeof(var_array) / sizeof(var_array[0]))))) { int x = 0; for (x = 0; x < var_count; x++) { char *inner_var_array[2] = { 0 }; int inner_var_count; if ((inner_var_count = switch_separate_string(var_array[x], '=', inner_var_array, (sizeof(inner_var_array) / sizeof(inner_var_array[0])))) == 2) { switch_channel_set_variable(originate_status[i].peer_channel, inner_var_array[0], inner_var_array[1]); } } } } if (!local_clobber) { if (var_event) { switch_event_t *event; switch_event_header_t *header; /* install the vars from the {} params */ for (header = var_event->headers; header; header = header->next) { switch_channel_set_variable(originate_status[i].peer_channel, header->name, header->value); } switch_event_create(&event, SWITCH_EVENT_CHANNEL_ORIGINATE); switch_assert(event); switch_channel_event_set_data(originate_status[i].peer_channel, event); switch_event_fire(&event); } } The code block inside if (local_clobber) and if (!local_clobber) does exactly the same thing. May I know is it by design and how does it actually work ? I am not sure what does it mean by localClobber so if someone can spare sometime explaining, I would really appreciate it. Thank you Best Regards Alex To _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100127/6c246887/attachment-0001.html From devel at thom.fr.eu.org Wed Jan 27 00:48:16 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 27 Jan 2010 09:48:16 +0100 Subject: [Freeswitch-dev] Openzap and CLIP In-Reply-To: References: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> <2767a9f20c8cc35f4c1c62c43528f9b2@thom.fr.eu.org> Message-ID: <613822f67a7f60eb824c246d4be7fcec@thom.fr.eu.org> So I did modify mod_openzap to provide the od trace command to the cli. I started trying to parse the incoming clip, but I have to do it again as the incoming signal is saturated so I can't see anything interesting. I'll post my results whenever I get something. Shall I send the modified mod_openzap.c to the list ? About the difference on the modulation settings for CLIP, could anybody suggest some way to have it configurable by files ? Thanks Fran?ois On Wed, 20 Jan 2010 14:01:03 -0500, Moises Silva wrote: Hi again Fran?ois, Please keep the conversation in the freeswitch-dev mailing list, unless you want private paid support ;-) See the answer to your question below ... On Wed, Jan 20, 2010 at 3:57 AM, Fran?ois Legal wrote: Do you know if there is a way (except using an oscilloscope to sample the line signal) to get a dump of the data sent to the card by openzap (is it samples or something) and vice versa. What I would like to do is see what is returned to openzap when a call arrives on an FXO port (CLIP is working in that direction) and compare it to what is being sent to the card when sending CLIP to and FXS port. Once the call gets into the computer we're talking about digital audio samples. In a typical configuration a Sangoma board will give you 160 samples of audio each 20ms (most likely encoded in ulaw, that can be verified in /etc/wanpipe/wanpipex.conf for the FXO/FXS board, look at the TDMV_LAW parameter). There is an openzap API command to record the digital audio read from the board, check ZAP_COMMAND_TRACE_INPUT and ZAP_COMMAND_TRACE_OUTPUT. What we currently lack is a FreeSWITCH command in mod_openzap to enable/disable the audio tracing. It should be trivial to hack mod_openzap.c and add that command, if you need help with that ping me on IRC (moy). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com [2] Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100127/3dad64bb/attachment.html From mike at jerris.com Wed Jan 27 08:22:13 2010 From: mike at jerris.com (Michael Jerris) Date: Wed, 27 Jan 2010 11:22:13 -0500 Subject: [Freeswitch-dev] Openzap and CLIP In-Reply-To: <613822f67a7f60eb824c246d4be7fcec@thom.fr.eu.org> References: <000601ca993a$e4e18be0$aea4a3a0$@fr.eu.org> <2767a9f20c8cc35f4c1c62c43528f9b2@thom.fr.eu.org> <613822f67a7f60eb824c246d4be7fcec@thom.fr.eu.org> Message-ID: <6F87F7C2-5532-464A-AC86-BC0DC91E4356@jerris.com> On Jan 27, 2010, at 3:48 AM, Fran?ois Legal wrote: > So I did modify mod_openzap to provide the od trace command to the cli. I started trying to parse the incoming clip, but I have to do it again as the incoming signal is saturated so I can't see anything interesting. > > > I'll post my results whenever I get something. > > > Shall I send the modified mod_openzap.c to the list ? > > We should get a bug opened on this issue on http://jira.freeswitch.org > About the difference on the modulation settings for CLIP, could anybody suggest some way to have it configurable by files ? > > We should add some config option based on country or CLIP type to specify this. Lets get it working and we can think how to configure once we have that. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100127/182d747a/attachment.html From m.sobkow at marketelsystems.com Wed Jan 27 11:37:40 2010 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 27 Jan 2010 13:37:40 -0600 Subject: [Freeswitch-dev] .deb sound installation files Message-ID: <4B609604.2010509@marketelsystems.com> It looks like the sound files are no longer part of the standard build, as per emails that were sent out some time ago. Is there any word on .deb support for building the sound files? Our install process requires that we do .deb files for ubuntu rather than doing a source build. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com From m.sobkow at marketelsystems.com Wed Jan 27 13:57:53 2010 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 27 Jan 2010 15:57:53 -0600 Subject: [Freeswitch-dev] .deb sound installation files In-Reply-To: <4B609604.2010509@marketelsystems.com> References: <4B609604.2010509@marketelsystems.com> Message-ID: <4B60B6E1.8050500@marketelsystems.com> Mark Sobkow wrote: > It looks like the sound files are no longer part of the standard > build, as per emails that were sent out some time ago. > > Is there any word on .deb support for building the sound files? Our > install process requires that we do .deb files for ubuntu rather than > doing a source build. > Is there any reason we can't use the .deb sound files from 1.0.4? -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com From edpimentl at gmail.com Wed Jan 27 16:23:34 2010 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 27 Jan 2010 19:23:34 -0500 Subject: [Freeswitch-dev] Call for help on FSComm In-Reply-To: References: Message-ID: <9dc4a1671001271623m1f0108c4seb4c7f0d87d64db7@mail.gmail.com> KUDOS!!! on FSComm... While searching for FsComm, I discover this link http://fscomm.com/ If you decide to consider alternative names for fscomm here are my suggestions: 1- FSSIP "FreeSwitch Services over IP" 2- FScIM "FreeSwitch Communication Interaction Management" 3- FS_IpCom "FreeSwitch Ip Communication" 4- FSiCA "FreeSwitch Intelligent Communication Agent' 5- uCA "Ubiquitous Communication Agent" 6- FComm "FreeSwitch Communication" 7- KallMe Sincerely, -E http://vCardCloud.com http://JustGoogl.Me http://AgileCODES.com (2D QrCode for VoIP) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100127/972f4179/attachment.html From anthony.minessale at gmail.com Wed Jan 27 17:05:23 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Jan 2010 19:05:23 -0600 Subject: [Freeswitch-dev] Call for help on FSComm In-Reply-To: <9dc4a1671001271623m1f0108c4seb4c7f0d87d64db7@mail.gmail.com> References: <9dc4a1671001271623m1f0108c4seb4c7f0d87d64db7@mail.gmail.com> Message-ID: <191c3a031001271705n656b5737lf00ace2565faa43a@mail.gmail.com> fscomm is a nickname for "FreeSWITCH Communicator" and is distributed with the FS source so it's not very important if there is a fscomm.com You probably just gave that fscomm.com site more hits than it has ever gotten with that post ;) On Wed, Jan 27, 2010 at 6:23 PM, EdPimentl wrote: > KUDOS!!! on FSComm... > > While searching for FsComm, I discover this link http://fscomm.com/ > > If you decide to consider alternative names for fscomm here are my > suggestions: > > 1- FSSIP "FreeSwitch Services over IP" > 2- FScIM "FreeSwitch Communication Interaction Management" > 3- FS_IpCom "FreeSwitch Ip Communication" > 4- FSiCA "FreeSwitch Intelligent Communication Agent' > 5- uCA "Ubiquitous Communication Agent" > 6- FComm "FreeSwitch Communication" > 7- KallMe > > Sincerely, > -E > http://vCardCloud.com > http://JustGoogl.Me > http://AgileCODES.com (2D QrCode for VoIP) > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100127/af9db81c/attachment.html From mike at jerris.com Wed Jan 27 20:08:17 2010 From: mike at jerris.com (Michael Jerris) Date: Wed, 27 Jan 2010 23:08:17 -0500 Subject: [Freeswitch-dev] .deb sound installation files In-Reply-To: <4B60B6E1.8050500@marketelsystems.com> References: <4B609604.2010509@marketelsystems.com> <4B60B6E1.8050500@marketelsystems.com> Message-ID: <82B746FC-9985-4107-AA26-9797F21CF3F0@jerris.com> I think those are not done yet. They don't really belong in the freeswitch tree as they are not tightly tied to the freeswitch version, just a dependency. There have been a few changes in sounds back from 1.0.4. I assume they are missing some. Mike On Jan 27, 2010, at 4:57 PM, Mark Sobkow wrote: > Mark Sobkow wrote: >> It looks like the sound files are no longer part of the standard >> build, as per emails that were sent out some time ago. >> >> Is there any word on .deb support for building the sound files? Our >> install process requires that we do .deb files for ubuntu rather than >> doing a source build. >> > Is there any reason we can't use the .deb sound files from 1.0.4? From tonhudung at gmail.com Thu Jan 28 06:10:03 2010 From: tonhudung at gmail.com (Alex To) Date: Thu, 28 Jan 2010 22:10:03 +0800 Subject: [Freeswitch-dev] Question about switch_cond_next() Message-ID: <001101caa023$99eff5a0$cdcfe0e0$@com> Hi, would someone clarify this for me please Is it right to say that switch_cond_next() does only one thing which is to put the current thread to sleep for a period of time (1 second) ? Is it equivalent to, let's say, Thread.Sleep(xxx); or it also does something else besides putting the current thead to sleep ? Is switch_cond_yield the same as switch_cond_next, except that switch_cond_yield accepts parameter as the input interval ? Thanks Regards Alex To -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100128/1ff19294/attachment.html From anthony.minessale at gmail.com Thu Jan 28 08:33:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 Jan 2010 10:33:27 -0600 Subject: [Freeswitch-dev] Question about switch_cond_next() In-Reply-To: <001101caa023$99eff5a0$cdcfe0e0$@com> References: <001101caa023$99eff5a0$cdcfe0e0$@com> Message-ID: <191c3a031001280833x16373bi911b6b04fcdbcb9c@mail.gmail.com> cond_next is equiv to cond_yield(1000) or 1 ms actual time will vary depending on OS and HW On Thu, Jan 28, 2010 at 8:10 AM, Alex To wrote: > Hi, would someone clarify this for me please > > > > Is it right to say that switch_cond_next() does only one thing which is to > put the current thread to sleep for a period of time (1 second) ? > > > > Is it equivalent to, let?s say, Thread.Sleep(xxx); or it also does > something else besides putting the current thead to sleep ? > > > > Is switch_cond_yield the same as switch_cond_next, except that > switch_cond_yield accepts parameter as the input interval ? > > > > Thanks > > > > Regards > > > > Alex To > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 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/20100128/f6b6a010/attachment.html From vipkilla at gmail.com Thu Jan 28 18:00:53 2010 From: vipkilla at gmail.com (vip killa) Date: Thu, 28 Jan 2010 21:00:53 -0500 Subject: [Freeswitch-dev] FSComm PortAudio error Message-ID: <957f61371001281800u6b0d3e6ex71393f4b6ce0ae75@mail.gmail.com> Hi, when i run FSComm on Vista 64-bit. I get an error message saying Error querying audio devices when i click on File -> Preferences. Then under PortAudio (in preferences) there are no device. Is there a specific OS that FSComm will work on? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100128/8d1f03bf/attachment.html From jmesquita at freeswitch.org Thu Jan 28 21:44:02 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 29 Jan 2010 02:44:02 -0300 Subject: [Freeswitch-dev] FSComm PortAudio error In-Reply-To: <957f61371001281800u6b0d3e6ex71393f4b6ce0ae75@mail.gmail.com> References: <957f61371001281800u6b0d3e6ex71393f4b6ce0ae75@mail.gmail.com> Message-ID: Can you please pastebin the logs? You can find them on $HOME/.fscomm/log/freeswitch.log W/o that, I can't tell you much. Jo?o Mesquita FSComm Developer On Thu, Jan 28, 2010 at 11:00 PM, vip killa wrote: > Hi, when i run FSComm on Vista 64-bit. I get an error message saying Error > querying audio devices when i click on File -> Preferences. Then under > PortAudio (in preferences) there are no device. Is there a specific OS that > FSComm will work on? > > _______________________________________________ > 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/20100129/f0478179/attachment.html From msc at freeswitch.org Fri Jan 29 06:53:50 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 29 Jan 2010 06:53:50 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Agenda Message-ID: <87f2f3b91001290653x417398aaj8900a4da9b1aa83b@mail.gmail.com> Greetings, This week's conference call agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29 Please add your items as the agenda is very light this week. We do have a few things to discuss, though, so please hop on and bring a friend! Talk to you all soon, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100129/51d1f2c6/attachment.html From tonhudung at gmail.com Fri Jan 29 07:03:04 2010 From: tonhudung at gmail.com (Alex To) Date: Fri, 29 Jan 2010 23:03:04 +0800 Subject: [Freeswitch-dev] Question about switch_cond_next() In-Reply-To: <191c3a031001280833x16373bi911b6b04fcdbcb9c@mail.gmail.com> References: <001101caa023$99eff5a0$cdcfe0e0$@com> <191c3a031001280833x16373bi911b6b04fcdbcb9c@mail.gmail.com> Message-ID: <002401caa0f4$2cb56430$86202c90$@com> Thank you, Anthony Alex From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, January 29, 2010 12:33 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Question about switch_cond_next() cond_next is equiv to cond_yield(1000) or 1 ms actual time will vary depending on OS and HW On Thu, Jan 28, 2010 at 8:10 AM, Alex To wrote: Hi, would someone clarify this for me please Is it right to say that switch_cond_next() does only one thing which is to put the current thread to sleep for a period of time (1 second) ? Is it equivalent to, let's say, Thread.Sleep(xxx); or it also does something else besides putting the current thead to sleep ? Is switch_cond_yield the same as switch_cond_next, except that switch_cond_yield accepts parameter as the input interval ? Thanks Regards Alex To _______________________________________________ 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 iax:guest at conference.freeswitch.org/888 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/20100129/b0aec96d/attachment-0001.html From vipkilla at gmail.com Fri Jan 29 07:54:57 2010 From: vipkilla at gmail.com (vip killa) Date: Fri, 29 Jan 2010 10:54:57 -0500 Subject: [Freeswitch-dev] FSComm PortAudio error In-Reply-To: References: <957f61371001281800u6b0d3e6ex71393f4b6ce0ae75@mail.gmail.com> Message-ID: <957f61371001290754n63a133b3vfc81f9f5e6e33a90@mail.gmail.com> http://pastebin.freeswitch.org/11991 2010/1/29 Jo?o Mesquita > Can you please pastebin the logs? > > You can find them on $HOME/.fscomm/log/freeswitch.log > > W/o that, I can't tell you much. > > Jo?o Mesquita > FSComm Developer > > > On Thu, Jan 28, 2010 at 11:00 PM, vip killa wrote: > >> Hi, when i run FSComm on Vista 64-bit. I get an error message saying Error >> querying audio devices when i click on File -> Preferences. Then under >> PortAudio (in preferences) there are no device. Is there a specific OS that >> FSComm will work on? >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100129/86e8f767/attachment.html From pawel at voiceworks.pl Fri Jan 29 08:00:16 2010 From: pawel at voiceworks.pl (=?utf-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Fri, 29 Jan 2010 17:00:16 +0100 Subject: [Freeswitch-dev] getting out of proxy media Message-ID: Hi, I have a stupid case with proxy media. (I need it so T.38 works, bypass is not an option). I set proxy_media to true, do a simple bridge to a sip destination and then, once it fails, I need to play a prompt that it failed. The bridge fails with SIP cause 404 after a second worth of G.729 early media (can't ignore the early media) Tried: a) pre_answer + playback b) answer + playback c) redirect to a sip destination with a proper prompt d) deflect to a sip destination with a proper prompt e) bridge to a sip destination with a proper prompt while forcing PCMA codec A & B fail with Codec PROXY PASS-THROUGH encoder error! C - hangs the channel indefinitely D - hangups the channel without sending SIP REDIRECT (call is never fully answered) E - fails with Codec PROXY PASS-THROUGH encoder error! Any ideas how can I enjoy proxy_media and yet be able to play anything to the user if the bridge setup fails ? Bounty? Pawel, -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3819 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100129/327ce244/attachment.bin From jmesquita at freeswitch.org Fri Jan 29 08:39:31 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 29 Jan 2010 14:39:31 -0200 Subject: [Freeswitch-dev] FSComm PortAudio error In-Reply-To: <957f61371001290754n63a133b3vfc81f9f5e6e33a90@mail.gmail.com> References: <957f61371001281800u6b0d3e6ex71393f4b6ce0ae75@mail.gmail.com> <957f61371001290754n63a133b3vfc81f9f5e6e33a90@mail.gmail.com> Message-ID: 2010-01-28 20:45:14.971000 [ERR] mod_PortAudio.c:978 Cannot find an input device Will make modifications to warn users but that means that portaudio did not find a mic or similar. Jo?o Mesquita FSComm Developer On Fri, Jan 29, 2010 at 1:54 PM, vip killa wrote: > http://pastebin.freeswitch.org/11991 > > 2010/1/29 Jo?o Mesquita > > Can you please pastebin the logs? >> >> You can find them on $HOME/.fscomm/log/freeswitch.log >> >> W/o that, I can't tell you much. >> >> Jo?o Mesquita >> FSComm Developer >> >> >> On Thu, Jan 28, 2010 at 11:00 PM, vip killa wrote: >> >>> Hi, when i run FSComm on Vista 64-bit. I get an error message saying >>> Error querying audio devices when i click on File -> Preferences. Then under >>> PortAudio (in preferences) there are no device. Is there a specific OS that >>> FSComm will work on? >>> >>> _______________________________________________ >>> 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 >> >> > > _______________________________________________ > 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/20100129/ff2f66f0/attachment.html From ustcorporation at yahoo.com Fri Jan 29 21:00:14 2010 From: ustcorporation at yahoo.com (Darren C.) Date: Fri, 29 Jan 2010 21:00:14 -0800 (PST) Subject: [Freeswitch-dev] Seeking Advice on SIP Phones like Aastra 6739i or Snom 870 that have good interoperability with FreeSWITCH Message-ID: <494371.30408.qm@web33004.mail.mud.yahoo.com> Hello, I'm working on a project and are interested in using one of the newer SIP Phones with color displays, perhaps touchscreen, etc. We want to send some information about a call to the SIP Phone either via FS or our own Web Service. These two phones have XML browsers that we may be able to utilize. I prefer to use phones that work well with FS but these are too new to show up on interoperability page. Anyone have experience using these, any advise would be appreciated: Aastra 6739i http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pds_en_1209.pdf http://www.voipsupply.com/aastra-6739i SNOM 870 http://www.snom.com/uploads/docu/data_snom870_en.pdf http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone/ http://www.voipsupply.com/snom-870 Thanks, teldev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100129/a64d44ad/attachment-0001.html From plite2012 at gmail.com Fri Jan 29 21:32:17 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 29 Jan 2010 23:32:17 -0600 Subject: [Freeswitch-dev] FSComm PortAudio error Message-ID: Most likely, your Vista 64-bit machine has a generic Vista 64-bit (High Definition) audio driver, instead of a specific audio driver for your machine's sound card. It seems PortAudio does not like the generic audio driver from Microsoft, even though you can play audio/video fine with Windows media player or Real Player. This is from my experience. I installed Windows 7 on my old PC. At first, PortAudio failed to pick up sound device before I installed proper audio driver. When I finally installed proper audio driver, PortAudio works perfectly. Good luck to find a proper audio driver to your OS! From plite2012 at gmail.com Fri Jan 29 21:45:05 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 29 Jan 2010 23:45:05 -0600 Subject: [Freeswitch-dev] FSComm capable of multiple lines? Message-ID: With FSComm, may you put a call on hold, while taking or making another call? How about multiple lines? Say, you have more than two incoming lines, you take one connection, while leaving the rest on MOH (Music on Hold). I think this is achievable when FreeSwitch runs as standalone. Thanks a lot for your attention. From plite2012 at gmail.com Fri Jan 29 21:50:27 2010 From: plite2012 at gmail.com (Paul Li) Date: Fri, 29 Jan 2010 23:50:27 -0600 Subject: [Freeswitch-dev] FSComm script folder Message-ID: While FreeSWITCH set switch_directories.script_dir to "scripts", FSComm set switch_directories.script_dir to "script". Shouldn't it be "scripts" to be consistent? From marketing at cluecon.com Fri Jan 29 22:47:45 2010 From: marketing at cluecon.com (Michael Collins) Date: Fri, 29 Jan 2010 22:47:45 -0800 Subject: [Freeswitch-dev] ClueCon MMX - Save the Date! Message-ID: <87f2f3b91001292247u5f058054yca8590bb9c39ae65@mail.gmail.com> ClueCon MMX (2010) will be here before you know it! Please mark your calendars: August 3-5, 2010. Start talking up ClueCon with your peers, coworkers, business owners, CEOs, potential sponsors, and anyone else you can think of. It's coming fast, so start getting ready now. Lots more information will be coming soon. Looking forward to seeing everyone this August! -ClueCon Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100129/6d47d3ce/attachment.html From mike at jerris.com Fri Jan 29 23:36:34 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 30 Jan 2010 02:36:34 -0500 Subject: [Freeswitch-dev] getting out of proxy media In-Reply-To: References: Message-ID: <3072D315-9636-4989-87AB-06D776386D44@jerris.com> Likely this will never work right with proxy media, the solution is probably to add more real support for t.38 passthrough in freeswitch without the need for proxy mode by handing it properly in the sdp negotiation, and then in the rtp stack, when it switches to t.38, put just the rtp stack into proxy mode (different than the channel in proxy). This is similar to how we handle video today. Mike On Jan 29, 2010, at 11:00 AM, Pawe? Pier?cionek wrote: > Hi, > > I have a stupid case with proxy media. (I need it so T.38 works, bypass is not an option). > > I set proxy_media to true, do a simple bridge to a sip destination and then, once it fails, I need to play a prompt that it failed. > > The bridge fails with SIP cause 404 after a second worth of G.729 early media (can't ignore the early media) > > Tried: > > a) pre_answer + playback > b) answer + playback > c) redirect to a sip destination with a proper prompt > d) deflect to a sip destination with a proper prompt > e) bridge to a sip destination with a proper prompt while forcing PCMA codec > > A & B fail with Codec PROXY PASS-THROUGH encoder error! > C - hangs the channel indefinitely > D - hangups the channel without sending SIP REDIRECT (call is never fully answered) > E - fails with Codec PROXY PASS-THROUGH encoder error! > > Any ideas how can I enjoy proxy_media and yet be able to play anything to the user if the bridge setup fails ? > > Bounty? > > Pawel,_______________________________________________ > 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 vipkilla at gmail.com Sat Jan 30 13:00:46 2010 From: vipkilla at gmail.com (vip killa) Date: Sat, 30 Jan 2010 16:00:46 -0500 Subject: [Freeswitch-dev] Call for help on FSComm In-Reply-To: <191c3a031001271705n656b5737lf00ace2565faa43a@mail.gmail.com> References: <9dc4a1671001271623m1f0108c4seb4c7f0d87d64db7@mail.gmail.com> <191c3a031001271705n656b5737lf00ace2565faa43a@mail.gmail.com> Message-ID: <957f61371001301300k96ab9a5o7239b361cdcc065a@mail.gmail.com> http://pastebin.freeswitch.org/12011 On Wed, Jan 27, 2010 at 8:05 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > fscomm is a nickname for "FreeSWITCH Communicator" and is distributed with > the FS source so it's not very important if there is a fscomm.com > > You probably just gave that fscomm.com site more hits than it has ever > gotten with that post ;) > > > On Wed, Jan 27, 2010 at 6:23 PM, EdPimentl wrote: > >> KUDOS!!! on FSComm... >> >> While searching for FsComm, I discover this link http://fscomm.com/ >> >> If you decide to consider alternative names for fscomm here are my >> suggestions: >> >> 1- FSSIP "FreeSwitch Services over IP" >> 2- FScIM "FreeSwitch Communication Interaction Management" >> 3- FS_IpCom "FreeSwitch Ip Communication" >> 4- FSiCA "FreeSwitch Intelligent Communication Agent' >> 5- uCA "Ubiquitous Communication Agent" >> 6- FComm "FreeSwitch Communication" >> 7- KallMe >> >> Sincerely, >> -E >> http://vCardCloud.com >> http://JustGoogl.Me >> http://AgileCODES.com (2D QrCode for VoIP) >> >> _______________________________________________ >> 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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/20100130/19b65fd4/attachment.html From db at bellette.net Sat Jan 30 14:59:57 2010 From: db at bellette.net (DB) Date: Sun, 31 Jan 2010 08:59:57 +1000 Subject: [Freeswitch-dev] Aborting playing of audio In-Reply-To: <957f61371001301300k96ab9a5o7239b361cdcc065a@mail.gmail.com> References: <9dc4a1671001271623m1f0108c4seb4c7f0d87d64db7@mail.gmail.com> <191c3a031001271705n656b5737lf00ace2565faa43a@mail.gmail.com> <957f61371001301300k96ab9a5o7239b361cdcc065a@mail.gmail.com> Message-ID: <002f01caa1ff$f2f4c3a0$d8de4ae0$@net> I was wondering if there is a way to abort the playback of an audio file on an external event? e.g. if a background thread was used to perform some long running calculation or remote data lookup, once the data was returned, is there a way I can abort the playback of a file currently being played? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... 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