February 2009 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] What's the easiest way to get all the SIP messages in FS?
by seven du 31 Mar '09

31 Mar '09

Hi, I want to get all(or certain) SIP messages in FS(perhaps I want to write a module), What's the easiest or best way to get that? or where I can find it in code? I know we can see SIP messages with TPORT_LOG enabled, It's in mod_sofia or sofia-lib? Why it is only possible to set that on start time but can't change it lively? If I can get them, I think it's possible to write to a file/db, or even send out from event_socket. I know it will hurt performance, but we don't do high volume calls and want to debug easily. Third party tools maybe work but also hurt performance right? Thanks. Seven

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[Freeswitch-dev] FS send re-invite about every minutes, is there a way to disable it?
by seven 01 Mar '09

01 Mar '09

Hi, The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description". I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it? btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.

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[Freeswitch-dev] can't set caller id opal/h323
by Sergio Alecha 24 Feb '09

24 Feb '09

they can set caller id in module mod_opal (opal/h323) ?. If possible, as is done? Thanks.

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Re: [Freeswitch-dev] Conference Hangup
by Michael S Collins 18 Feb '09

18 Feb '09

On Feb 17, 2009, at 9:07 PM, Scott Shen <sshen(a)oa.com.au> wrote: > Thanks for the quick response, however, I meant "group call", > something like > > <action application="conference_set_auto_outcall" data="$ > {group_call(sales)}"/> > > could I do something like > > <action application="conference_set_auto_outcall" data="$ > {group_call(sales)}+flag(endconf)"/> > > In this case, I would assume anyone in the group could terminate the > conference upon leaving, is that right? IIRC, group_call connects to a single endpoint. That is, it dials a group but whoever answers "wins" and the other phones stop ringing. Is that what you are thinking? -MC >> >> >> >> >> >> >>> >> >> >> >>

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[Freeswitch-dev] Conference Hangup
by Scott Shen 18 Feb '09

18 Feb '09

Is there a way to configure the dialplan so that the FS could terminate the conference session upon certain condition? The scenario is: 1. User A dials a number 2. FS starts a conference session by ringing User B and User C ( and User D or any number of other Users ) 3. User B joins the conference ( along with other invited Users ) 4. After talking a while, User B leaves the conference 5. At here, could FS be triggered to terminate the conference? Regards Scott

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[Freeswitch-dev] jira question
by Apostolos Pantsiopoulos 17 Feb '09

17 Feb '09

Hi, I have just created a second patch for module mod_radius_cdr which is a better version of my previous patch : http://jira.freeswitch.org/browse/MODEVENT-38?page=com.atlassian.jira.plugi… What I would like to know is the procedure that is used by the maintainers to test (and hopefully approve) the submitted patches. I cloned MODEVENT-30 because it was relevant to the issue that I tried to solve. The system automatically set Chris Parker as the assignee. So, where does it go from here? Do I wait for the patch to be reviewed by the assignee or the project leader? I am not in any specific hurry about it, I am just wondering how the procedure goes... -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Re: [Freeswitch-dev] [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH
by Giovanni Maruzzelli 16 Feb '09

16 Feb '09

On Sun, Feb 15, 2009 at 2:45 PM, Gourav Shah <freedomyug(a)gmail.com> wrote: > I have been testing it for a couple of days now, and its been working fine > in my environment for making outgoing calls through skypeout etc. Have > filed a ticket for you regarding incoming call transfer to regular > extensions. (http://jira.freeswitch.org/browse/MODSKYPIAX-4) I tested and it worked for me. Maybe was the xml bug fixed by anthm? Please "make current" FS, or reinstall from scratch, and check again... If the problem show up again, reopen the issue. > > I see you have been planning to release a Live CD for Celliax for some time > now, and would like to offer my help with this. If you need a resource to > create a Live CD, please let me know, I would be glad to help. I would be glad to collaborate on this. The livecd has been neglected until now, and Celliax too, because of Skypiax ;-). Starting monday I'll begin to port Celliax to FS. Would be nice to have the livecd with both Celliax and Skypiax, on both FS and *. In the past I used Knoppix as a base for the livecds, maybe today we can start from Ubuntu live cd? What do you think? Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Feb 15, 2009 at 2:45 PM, Gourav Shah <freedomyug(a)gmail.com> wrote: > Thank You ! > > I have been waiting for mod_skypiax to be released on FreeSwitch. > > I have been testing it for a couple of days now, and its been working fine > in my environment for making outgoing calls through skypeout etc. Have > filed a ticket for you regarding incoming call transfer to regular > extensions. (http://jira.freeswitch.org/browse/MODSKYPIAX-4) > > I see you have been planning to release a Live CD for Celliax for some time > now, and would like to offer my help with this. If you need a resource to > create a Live CD, please let me know, I would be glad to help. > > Thanks > Gourav > > On Fri, Feb 13, 2009 at 11:45 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org> > wrote: >> >> Hello FreeSWITCHers, >> >> mod_skypiax is available for testing, feature requests, bug hunting. >> >> I would like to ask the help of you all to make Skypiax robust and >> feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX >> on IRC), that has developed mod_airpe (another Skype endpoint). >> >> I've written a first documentation on Skypiax installation and usage >> at: http://wiki.freeswitch.org/wiki/Skypiax >> and there is a Jira module at: >> http://jira.freeswitch.org/browse/MODSKYPIAX >> >> So, please, test the software, edit the wiki page both for style and >> content, file bug reports and feature requests. >> >> FreeSWITCH is now the platform of first development for me, so the >> FreeSWITCH part of Skypiax is more tested (if any) and the code is >> more readable compared to the Asterisk part where lot of legacy from >> my other projects clutter the code. >> >> But Skypiax strive to be available as a Skype compatible endpoint for >> all the opensource telephony community, and in the near time the >> Asterisk part will be cleaned much more, and documented. >> >> As you will see, the code is made by skypiax_protocol.c (the >> interaction with Skype client), mod_skypiax.c (the interaction with >> FreeSWITCH), chan_skypiax.c (the interaction with Asterisk). >> >> Please consider me available for all infos, clarifications, discussions, >> etc. >> >> I would like to thanks all the peoples that helped me via mail and IRC >> (so bad to have different timezones, isn't?), the *very early >> adopters*, the testers, the patchers, and you all. >> Particularly Anthony Minessale, Michael Jerris, Brian West, Michael >> Collins, Ken Rice, Seven Du, Clif Cox, Hristo Trendev, Rehan Allah >> Wala, Jason Garland and Antonio Gallo. >> >> >From the wiki page (http://wiki.freeswitch.org/wiki/Skypiax) >> >> WHAT IS SKYPIAX >> >> This software (Skypiax) uses the Skype API but is not endorsed, >> certified or otherwise approved in any way by Skype. >> >> Skypiax is an endpoint (channel driver) that uses the Skype client as >> an interface to the Skype network, and allows incoming and outgoing >> Skype calls to/from FreeSWITCH (that can be bridged, originated, >> answered, etc. as in all other endpoints, e.g. sofia/SIP). >> >> Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both >> 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on >> Asterisk too, at 8khz, on Linux and Windows (through CygWin). >> >> Think of Skypiax as similar to OpenZAP for analog lines. For each >> channel you need an interface (a Skype client). So, for example, two >> concurrent calls would need two channels, and therefor two Skype >> clients running on your FreeSWITCH server. >> >> If your Skype client(s) have Skype credits, then Skypiax works for >> SkypeOut calls as well. >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Thu, Jan 15, 2009 at 6:39 PM, Giovanni Maruzzelli >> <gmaruzz(a)celliax.org> wrote: >> > Hi FreeSWITCH developers >> > >> > I would like to propose to the community my plans, so we can discuss >> > and coordinate efforts. >> > >> > I developed a couple of channel drivers for Asterisk in the past >> > (works on both Linux and Windows), and I would like to port them to FS >> > and further enhance them. >> > >> > The two endpoints are: >> > - Skypiax, Skype compatible, makes and receives calls to/from Skype >> > network and Skypeout service, using the Skype client as interface. >> > - Celliax, GSM and SMS endpoint, makes and receives voice calls and >> > SMSs to/from the GSM/CDMA network, using second hand cellphones and/or >> > embedded professional devices as interfaces >> > >> > My aims are: >> > >> > a) port both endpoints from Asterisk to FreeSWITCH >> > b) have both endpoints continue to support at least Linux and Windows on >> > FS >> > c) I would like better having most of the endpoints code working for >> > both FreeSWITCH and Asterisk, maintaining separated the code that >> > interface with the GSM and Skype network, from the code that interface >> > with the core. >> > >> > Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM >> > endpoint, and they share the same skeleton and logic, so porting >> > celliax after having ported skypiax will be easier and faster :-). >> > >> > Current situation and next steps: >> > 1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available >> > for testing and debugging, needs to be polished and cleaned >> > 2) starting mid next week (I'll be back in office), I want to >> > integrate into skypiax the code and ideas from mod_airpe of Massimo >> > (ctrix), that has developed an alternative Skype compatible module, >> > and coordinate any future development with him and any other >> > interested developer >> > 3) begin the porting of celliax to FS, aiming at a pre-beta release >> > for Linux and Windows during February. >> > 4) coordinate further development of celliax with any other developer >> > interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands, >> > embedded devices, audio sampling >> > >> > I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more >> > info at www.celliax.org. >> > >> > Sincerely, >> > >> > Giovanni Maruzzelli >> > ========================================= >> > www.celliax.org >> > via Pierlombardo 9, 20135 Milano >> > Italy >> > gmaruzz at celliax dot org >> > Cell : +39-347-2665618 >> > Fax : +39-02-87390039 >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > >

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[Freeswitch-dev] FreeSWITCH Windows Crash due to Detached Threads (and solution)
by Ron Avriel 16 Feb '09

16 Feb '09

Hi, Under Windows, FreeSWITCH consistently crashes with all default install setting after issuing the ‘shutdown’ command from console immediately after startup, without a single call handled. Note that on my servers, this only happens in ‘Release’ version, not in ‘Debug’. The problem is due to “problematic” Windows thread handling. I noticed that all FS threads are ‘detachable’ and not ‘joinable’. When an application with active detached threads terminates, a race condition can lead to the application crash, as shown in the sample application at http://pastebin.freeswitch.org/7225 . This is exactly what causes FS to crash as a result of the mod_event_socket detached runtime thread. A temporary workaround is to change in switch_loadable_module_exec() the call to switch_yield(1000000) with switch_thread_exit(thread, 0). In my case it solved the crash, but theoretically it does not always solve the problem, because there is no guarantee that it will get called before the main thread reaches its termination code. I think the only safe solution is to use ‘joinable’ threads. Is there a good reason they’re not used? Note – this problem was reported to jira about a month ago http://jira.freeswitch.org/browse/MODEVENT-37 but it seems it was ignored. Thanks, Ron

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[Freeswitch-dev] Ecasound support in FreeSWITCH?
by Kristian Kielhofner 13 Feb '09

13 Feb '09

Hello everyone, I started a project using Asterisk + Ecasound to do call quality analysis: http://www.voip-info.org/wiki/view/Recqual I'd like to expand on this concept (that's about all there is). I just don't have as much time to spend on this as I'd like. It seems to me that Ecasound could be built into FreeSWITCH as a mod: http://ecasound.seul.org/ecasound/developer.php What I'm thinking is FreeSWITCH could dial a remote resource and give some params to mod_ecasound with the quality/type of audio it is expecting. Mod_ecasound could report what it's getting. For instance, you could dial any standard milliwatt test line and mod_ecasound would tell you how mangled the received audio is. Obviously you could also call your own resources, loop back into other areas of the dialplan, etc. I don't know how well suited ecasound is to doing this kind of analysis in real time or if it's the best way of going about this but it does seem like FreeSWITCH is an ideal platform for any future recqual development. Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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[Freeswitch-dev] Skypiax (Skype Network endpoint) for FreeSWITCH
by Giovanni Maruzzelli 13 Feb '09

13 Feb '09

Hello FreeSWITCHers, mod_skypiax is available for testing, feature requests, bug hunting. I would like to ask the help of you all to make Skypiax robust and feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX on IRC), that has developed mod_airpe (another Skype endpoint). I've written a first documentation on Skypiax installation and usage at: http://wiki.freeswitch.org/wiki/Skypiax and there is a Jira module at: http://jira.freeswitch.org/browse/MODSKYPIAX So, please, test the software, edit the wiki page both for style and content, file bug reports and feature requests. FreeSWITCH is now the platform of first development for me, so the FreeSWITCH part of Skypiax is more tested (if any) and the code is more readable compared to the Asterisk part where lot of legacy from my other projects clutter the code. But Skypiax strive to be available as a Skype compatible endpoint for all the opensource telephony community, and in the near time the Asterisk part will be cleaned much more, and documented. As you will see, the code is made by skypiax_protocol.c (the interaction with Skype client), mod_skypiax.c (the interaction with FreeSWITCH), chan_skypiax.c (the interaction with Asterisk). Please consider me available for all infos, clarifications, discussions, etc. I would like to thanks all the peoples that helped me via mail and IRC (so bad to have different timezones, isn't?), the *very early adopters*, the testers, the patchers, and you all. Particularly Anthony Minessale, Michael Jerris, Brian West, Michael Collins, Ken Rice, Seven Du, Clif Cox, Hristo Trendev, Rehan Allah Wala, Jason Garland and Antonio Gallo. >From the wiki page (http://wiki.freeswitch.org/wiki/Skypiax) WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that uses the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls to/from FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, e.g. sofia/SIP). Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin). Think of Skypiax as similar to OpenZAP for analog lines. For each channel you need an interface (a Skype client). So, for example, two concurrent calls would need two channels, and therefor two Skype clients running on your FreeSWITCH server. If your Skype client(s) have Skype credits, then Skypiax works for SkypeOut calls as well. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Jan 15, 2009 at 6:39 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org> wrote: > Hi FreeSWITCH developers > > I would like to propose to the community my plans, so we can discuss > and coordinate efforts. > > I developed a couple of channel drivers for Asterisk in the past > (works on both Linux and Windows), and I would like to port them to FS > and further enhance them. > > The two endpoints are: > - Skypiax, Skype compatible, makes and receives calls to/from Skype > network and Skypeout service, using the Skype client as interface. > - Celliax, GSM and SMS endpoint, makes and receives voice calls and > SMSs to/from the GSM/CDMA network, using second hand cellphones and/or > embedded professional devices as interfaces > > My aims are: > > a) port both endpoints from Asterisk to FreeSWITCH > b) have both endpoints continue to support at least Linux and Windows on FS > c) I would like better having most of the endpoints code working for > both FreeSWITCH and Asterisk, maintaining separated the code that > interface with the GSM and Skype network, from the code that interface > with the core. > > Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM > endpoint, and they share the same skeleton and logic, so porting > celliax after having ported skypiax will be easier and faster :-). > > Current situation and next steps: > 1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available > for testing and debugging, needs to be polished and cleaned > 2) starting mid next week (I'll be back in office), I want to > integrate into skypiax the code and ideas from mod_airpe of Massimo > (ctrix), that has developed an alternative Skype compatible module, > and coordinate any future development with him and any other > interested developer > 3) begin the porting of celliax to FS, aiming at a pre-beta release > for Linux and Windows during February. > 4) coordinate further development of celliax with any other developer > interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands, > embedded devices, audio sampling > > I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more > info at www.celliax.org. > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 >

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