November 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Minor log cleanup in sofia_reg.c
by Beeton, Carolyn (Carolyn) 11 Nov '10

11 Nov '10

Here's a patch for a minor log cleanup in sofia_reg.c. The "forbidden" variable is never set, so the logs keep showing a "challenge" being sent instead of "forbidden", when a registration is rejected. diff --git a/src/mod/endpoints/mod_sofia/sofia_reg.c b/src/mod/endpoints/mod_sofia/sofia_reg.c index 67641ae..61bb4aa 100644 --- a/src/mod/endpoints/mod_sofia/sofia_reg.c +++ b/src/mod/endpoints/mod_sofia/sofia_reg.c @@ -816,7 +816,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand char contact_str[1024] = ""; int nat_hack = 0; uint8_t multi_reg = 0, multi_reg_contact = 0, avoid_multi_reg = 0; - uint8_t stale = 0, forbidden = 0; + uint8_t stale = 0; auth_res_t auth_res; long exptime = 300; switch_event_t *event; @@ -1115,7 +1115,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand if (auth_res != AUTH_OK && !stale) { if (profile->debug) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send %s for [%s@%s]\n", forbidden ? "forbidden" : "challenge", to_user, to_host); + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send %s for [%s@%s]\n", (auth_res == AUTH_FORBIDDEN) ? "forbidden" : "challenge", to_user, to_host); } if (auth_res == AUTH_FORBIDDEN) { nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());

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[Freeswitch-dev] Announcing the Jester Mail project
by Chad Phillips -- Apartment Lines 11 Nov '10

11 Nov '10

With funding and support from Star2Star Communications, and in conjunction with FreeSWITCH Solutions, I will soon be embarking on a project to bring a drop-in replacement for Asterisk's Comedian Mail to the FreeSWITCH platform. As I learned in my own move from Asterisk to FreeSWITCH, one of the bigger challenges was finding a transparent way to transition my user's voicemail experience. Since FreeSWITCH's native voicemail module did not adequately meet this need, I elected to write my own voicemail system, leveraging Lua -- the recommended scripting language for integrating complex tasks into the dialplan. The underlying concept in Jester Mail is to provide a flexible, extensible, and easily customizable voicemail plugin for FreeSWITCH. Unlike mod_voicemail, the code/configuration will be written in either XML or Lua, so that it will be easily hackable should the need arise. Every effort will be made to make the components modular and configurable, to avoid the need to rewrite any of the core functionality. Initial alpha releases will be based on my existing company's code, adding more and more of the standard Comedian Mail features, culminating in the 1.0 release, which will be a complete drop in replacement for Comedian Mail. Future releases may extend functionality even beyond that provided in Comedian Mail -- we'll see about that when we get there. :) We'd like to encourage people to offer feedback and insight on the project as it progresses, and we'll do our best (as it fits into our budget, timeline, and larger goals) to incorporate this into our final product in a way that is most beneficial for all in the community. Within the next 2-3 weeks, I'll be posting a strategy/architecture battle plan on the FreeSWITCH wiki. A few weeks after that I'll commit the initial pre-alpha code to the contributions repository for community feedback. There should be two nice side effects of this project: a) More people should be willing to make the transition from Asterisk to FreeSWITCH, as this removes another significant barrier for some. b) We will have excellent new resource to point people to for learning how to leverage Lua in FreeSWITCH. Chad Phillips aka hunmonk

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 10 Nov '10

10 Nov '10

Hey folks, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_10 We have a special announcement from Chad Philips and Kristian Kielhofner so be sure to join us today! -Michael

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[Freeswitch-dev] Getting DTMF from channel.
by Paulo Rogério Panhoto 09 Nov '10

09 Nov '10

Hi, I'm writing a module that allows playback of MP4 video files (with libmp4v2). The playback itself is made by two functions ready to run on separate threads (though, audio runs on current thread and video runs on a separate one). At this point, I'm trying to implement dtmf cut-through. After some research (I checked out mod_dptools.c and switch_ivr_play_say.c) and this code was my best guess -- it runs on the audio stream: if(switch_channel_test_flag(pt->channel, CF_BREAK)) { switch_channel_clear_flag(pt->channel, CF_BREAK); break; } switch_ivr_parse_all_events(pt->session); if(switch_channel_has_dtmf(pt->channel)) { switch_channel_dequeue_dtmf(pt->channel, &dtmf); const char * terminators = switch_channel_get_variable(pt->channel, SWITCH_PLAYBACK_TERMINATORS_VARIABLE); if(terminators && !strcasecmp(terminators, "none")) terminators = NULL; switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(pt->session), SWITCH_LOG_DEBUG, "Digit %c\n", dtmf.digit); if(terminators && strchr(terminators, dtmf.digit)) { std::string digit(&dtmf.digit, 0, 1); switch_channel_set_variable(pt->channel, SWITCH_PLAYBACK_TERMINATOR_USED, digit.c_str()); break; } } Which didn't work. I'm asking if anyone has any idea. Any help is appreciated Regards, Paulo R. Panhoto

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Re: [Freeswitch-dev] Session Progress and RINGING
by Bernhard Suttner 05 Nov '10

05 Nov '10

We will try that out but its not "100%" the functionality as the sbc does signalize. If I understand you correct, the 183 with SDP will be forwarded to A with 180 with SDP. But this does change the functionality to A (first of all on SIP/ISDN gateways). Therefore I would prefer the "clean" way with RFC 3960. Would you include a patch like "support-rfc3960" configuration option to sofia config which will ignore the CF_EARLY_MEDIA test? ----- Original Message ----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] To: freeswitch-dev(a)lists.freeswitch.org Sent: Thu, 04 Nov 2010 22:54:34 +0100 Subject: Re: [Freeswitch-dev] Session Progress and RINGING > how about what I said: > {ignore_early_media=ring_ready} > > > > On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > yes, they are not really sure but I think that the given RFC does specify > this correctly (this does not mean, that all the SIP devices work like > that). I could understand that a change will maybe result in big troubles. > Perhaps a option for mod_sofia to "support-rfc3960" would be a good > solution. If you want I will write the patch because it should be "really > simple" to ignore the CF_EARLY_MEDIA if the option is set, or? > > > > We get the Session Progress with SDP and later the 180 Ringing from a > session border controller (I am not 100% sure, but I think its a a > Audiocodes). The 180 Ringing has to be sent towards A because A is a ISDN > gateway. > > > > Would do you prefer? Thanks for your investigation. > > > > Best regards, > > Bernhard Suttner > > > > ----- Original Message ----- > > From: Anthony Minessale [mailto:anthony.minessale@gmail.com] > > To: freeswitch-dev(a)lists.freeswitch.org > > Sent: Thu, 04 Nov 2010 22:09:48 +0100 > > Subject: Re: [Freeswitch-dev] Session Progress and RINGING > > > > > >> My conclusion from reading that discussion is they have no idea what to > do. > >> There are votes both ways and 2 attempts to go of onto a tangent into > >> another topic. > >> > >> We have chosen on our implementation to ignore 180 once we have > >> already established a media path from a 183. The RFC is just vague > >> enough that this decision falls to us. > >> > >> I understand that ISDN has more precise signaling in this regard. > >> PROGRESS or ALERTING both with or without media as a flag on the packet. > >> > >> But the vast majority of SIP devices in the wild will ignore the 180 > >> once early media has been established. > >> So even if we make changes to support this, it will be ignored by the > >> next guy in line. > >> > >> Do you hear media during this early media phase? > >> > >> This is why we have the originate param > >> {ignore_early_media=ring_ready} which will, in the case of sip, > >> translate 183 or 180 with and without SDP into a 180 > >> > >> To change it to do what you are asking for could have extreme negative > >> side effects and not even work in most devices so this is why I am > >> reluctant to change it. > >> > >> Are you trying to cross connect 2 ISDN lines over SIP and preserve the > >> signalling? > >> If so, this is why SIP is flawed to begin with because it is lossy in > >> telephony signaling data. > >> This is why they now try to embed ss7 messages in the sip packets. =p > >> > >> > >> > >> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner > >> <bernhard.suttner(a)winet.ch> wrote: > >> > Hi, > >> > > >> > session progress does have SDP which will then go through to A. But on > the > >> display of A it will only display "session progress" and not Ringing. If > A > >> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". > I > >> am not sure, if RINGING will be used for A to generate the ringback tone > for > >> itself on every device. This behaves on the client/device configuration. > >> > > >> > There was a nice discussion about this "issue" on the sip-implementors > >> list: > >> > > >> > > >> > http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg06400… > >> > > >> > There is a RFC for this case (Section: 3.2. Ringing Tone Generation) > >> > http://www.rfc-editor.org/rfc/rfc3960.txt > >> > > >> > What do you think? > >> > > >> > Best regards, > >> > Bernhard Suttner > >> > > >> > > >> > > >> > ----- Original Message ----- > >> > From: Anthony Minessale [mailto:anthony.minessale@gmail.com] > >> > To: freeswitch-dev(a)lists.freeswitch.org > >> > Sent: Thu, 04 Nov 2010 16:54:30 +0100 > >> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING > >> > > >> > > >> >> no, once it gets session progress it will not send any ringing. > >> >> The sip side 180 ringing is used to tell the phone to generate its own > >> >> inband ringing. > >> >> > >> >> does your 183 session progress contain a sdp? > >> >> > >> >> > >> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner > >> >> <bernhard.suttner(a)winet.ch> wrote: > >> >> > Hi, > >> >> > > >> >> > A ---> FS ---> B > >> >> > > >> >> > B does send a Session Progress to FS. FS will forward the Session > >> Progress > >> >> to A. > >> >> > B does send a RINGING to FS. FS does _not_ forward this to A. > >> >> > > >> >> > Could it be, that the check the switch_test_flag(channel, > >> CF_EARLY_MEDIA) > >> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think > it > >> is > >> >> allowed to send the Session Progress first and later the RINGING. The > >> >> ringing is for example for SIP/ISDN gateways necessary. > >> >> > > >> >> > Any hint is appreciated. > >> >> > > >> >> > Best regards, > >> >> > Bernhard Suttner > >> >> > > >> >> > > >> >> > > >> >> > _______________________________________________ > >> >> > FreeSWITCH-dev mailing list > >> >> > FreeSWITCH-dev(a)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@hotmail.com > >> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > >> >> IRC: irc.freenode.net #freeswitch > >> >> > >> >> FreeSWITCH Developer Conference > >> >> sip:888@conference.freeswitch.org > >> >> googletalk:conf+888@conference.freeswitch.org > >> >> pstn:+19193869900 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-dev mailing list > >> >> FreeSWITCH-dev(a)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(a)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@hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888@conference.freeswitch.org > >> googletalk:conf+888@conference.freeswitch.org > >> pstn:+19193869900 > >> > >> _______________________________________________ > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev(a)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(a)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@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org >

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Re: [Freeswitch-dev] Session Progress and RINGING
by Bernhard Suttner 04 Nov '10

04 Nov '10

Hi, yes, they are not really sure but I think that the given RFC does specify this correctly (this does not mean, that all the SIP devices work like that). I could understand that a change will maybe result in big troubles. Perhaps a option for mod_sofia to "support-rfc3960" would be a good solution. If you want I will write the patch because it should be "really simple" to ignore the CF_EARLY_MEDIA if the option is set, or? We get the Session Progress with SDP and later the 180 Ringing from a session border controller (I am not 100% sure, but I think its a a Audiocodes). The 180 Ringing has to be sent towards A because A is a ISDN gateway. Would do you prefer? Thanks for your investigation. Best regards, Bernhard Suttner ----- Original Message ----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] To: freeswitch-dev(a)lists.freeswitch.org Sent: Thu, 04 Nov 2010 22:09:48 +0100 Subject: Re: [Freeswitch-dev] Session Progress and RINGING > My conclusion from reading that discussion is they have no idea what to do. > There are votes both ways and 2 attempts to go of onto a tangent into > another topic. > > We have chosen on our implementation to ignore 180 once we have > already established a media path from a 183. The RFC is just vague > enough that this decision falls to us. > > I understand that ISDN has more precise signaling in this regard. > PROGRESS or ALERTING both with or without media as a flag on the packet. > > But the vast majority of SIP devices in the wild will ignore the 180 > once early media has been established. > So even if we make changes to support this, it will be ignored by the > next guy in line. > > Do you hear media during this early media phase? > > This is why we have the originate param > {ignore_early_media=ring_ready} which will, in the case of sip, > translate 183 or 180 with and without SDP into a 180 > > To change it to do what you are asking for could have extreme negative > side effects and not even work in most devices so this is why I am > reluctant to change it. > > Are you trying to cross connect 2 ISDN lines over SIP and preserve the > signalling? > If so, this is why SIP is flawed to begin with because it is lossy in > telephony signaling data. > This is why they now try to embed ss7 messages in the sip packets. =p > > > > On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > session progress does have SDP which will then go through to A. But on the > display of A it will only display "session progress" and not Ringing. If A > is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I > am not sure, if RINGING will be used for A to generate the ringback tone for > itself on every device. This behaves on the client/device configuration. > > > > There was a nice discussion about this "issue" on the sip-implementors > list: > > > > > http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg06400… > > > > There is a RFC for this case (Section: 3.2. Ringing Tone Generation) > > http://www.rfc-editor.org/rfc/rfc3960.txt > > > > What do you think? > > > > Best regards, > > Bernhard Suttner > > > > > > > > ----- Original Message ----- > > From: Anthony Minessale [mailto:anthony.minessale@gmail.com] > > To: freeswitch-dev(a)lists.freeswitch.org > > Sent: Thu, 04 Nov 2010 16:54:30 +0100 > > Subject: Re: [Freeswitch-dev] Session Progress and RINGING > > > > > >> no, once it gets session progress it will not send any ringing. > >> The sip side 180 ringing is used to tell the phone to generate its own > >> inband ringing. > >> > >> does your 183 session progress contain a sdp? > >> > >> > >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner > >> <bernhard.suttner(a)winet.ch> wrote: > >> > Hi, > >> > > >> > A ---> FS ---> B > >> > > >> > B does send a Session Progress to FS. FS will forward the Session > Progress > >> to A. > >> > B does send a RINGING to FS. FS does _not_ forward this to A. > >> > > >> > Could it be, that the check the switch_test_flag(channel, > CF_EARLY_MEDIA) > >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it > is > >> allowed to send the Session Progress first and later the RINGING. The > >> ringing is for example for SIP/ISDN gateways necessary. > >> > > >> > Any hint is appreciated. > >> > > >> > Best regards, > >> > Bernhard Suttner > >> > > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-dev mailing list > >> > FreeSWITCH-dev(a)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@hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888@conference.freeswitch.org > >> googletalk:conf+888@conference.freeswitch.org > >> pstn:+19193869900 > >> > >> _______________________________________________ > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev(a)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(a)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@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org >

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Re: [Freeswitch-dev] Session Progress and RINGING
by Bernhard Suttner 04 Nov '10

04 Nov '10

Hi, session progress does have SDP which will then go through to A. But on the display of A it will only display "session progress" and not Ringing. If A is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I am not sure, if RINGING will be used for A to generate the ringback tone for itself on every device. This behaves on the client/device configuration. There was a nice discussion about this "issue" on the sip-implementors list: http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg06400… There is a RFC for this case (Section: 3.2. Ringing Tone Generation) http://www.rfc-editor.org/rfc/rfc3960.txt What do you think? Best regards, Bernhard Suttner ----- Original Message ----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] To: freeswitch-dev(a)lists.freeswitch.org Sent: Thu, 04 Nov 2010 16:54:30 +0100 Subject: Re: [Freeswitch-dev] Session Progress and RINGING > no, once it gets session progress it will not send any ringing. > The sip side 180 ringing is used to tell the phone to generate its own > inband ringing. > > does your 183 session progress contain a sdp? > > > On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > A ---> FS ---> B > > > > B does send a Session Progress to FS. FS will forward the Session Progress > to A. > > B does send a RINGING to FS. FS does _not_ forward this to A. > > > > Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA) > in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is > allowed to send the Session Progress first and later the RINGING. The > ringing is for example for SIP/ISDN gateways necessary. > > > > Any hint is appreciated. > > > > Best regards, > > Bernhard Suttner > > > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev(a)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@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org >

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[Freeswitch-dev] CFP - Open Source Telephony Devroom - FOSDEM 2011
by Bryan M. Johns 04 Nov '10

04 Nov '10

Greetings, This is a call for presentations for the open source telephony devroom at FOSDEM 2011 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony on Sunday, February 6th. The schedule allows for presentations from 9:00 to 17:00. The room we have available will have a projector, wifi, and 59 seats. Please submit all proposals no later than 2010-12-10. Notification of accepted speakers will be provided by 2010-12-17. We will then work to have a schedule finalized by 2011-01-07. Talks should be submitted by fulling out the following form: http://goo.gl/9SWlz Please direct discussion about this devroom to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Russell Bryant <russell(a)digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! Bryan M. Johns Digium, Inc. | Community Director 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6007 Check us out at : www.asterisk.org or www.digium.com

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[Freeswitch-dev] Session Progress and RINGING
by Bernhard Suttner 04 Nov '10

04 Nov '10

Hi, A ---> FS ---> B B does send a Session Progress to FS. FS will forward the Session Progress to A. B does send a RINGING to FS. FS does _not_ forward this to A. Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA) in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is allowed to send the Session Progress first and later the RINGING. The ringing is for example for SIP/ISDN gateways necessary. Any hint is appreciated. Best regards, Bernhard Suttner

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[Freeswitch-dev] FreeSWITCH Conference Call Today - DRK Discusses .NET Fun
by Michael Collins 03 Nov '10

03 Nov '10

Hello all, The FreeSWITCH conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_03 Dave Kompel will be showing off some more .NET stuff that he's got cooking plus we have a few other things to discuss. Talk to you soon, Michael

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