June 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Broadvoice
by Jonathan Rogers 17 Jun '07

17 Jun '07

Has anyone tried to use FreeSwitch with Broadvoice? I'm having no luck getting it to authenticate. -- Jonathan Rogers <JonathanRRogers(a)gmail.com>

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A patent list for codecs
by Steve Underwood 17 Jun '07

17 Jun '07

Hi all, There is an issue which is common ground for most free telephony projects - codec patents. In the near term we need to live within the constraints of these patents, but they won't last forever (well, not as long as Disney isn't a big patent holder). The snag is working out when they will be free. There are several relevant encumbered speech codecs, potentially useful for what we do. The obvious ones are AMR-NB, AMR-WB, G.722.1, G.722.2 (which is basically AMR-WB by another name), G.723.1, G.728, G.729, G.729.1. If you look through the ETSI and ITU web sites you can find some material about who claims patents related to these codecs. However, the information is far from complete. In most cases the actual patent numbers claimed to be relevant are not listed. The dates of filing, granting and expiration are not generally listed, though if the patent numbers are available the dates are easy to find. I think it would be valuable to start a web page where all this information is gathered. We might then get a real picture of when a codec will become unencumbered. Clearly, video and even image, codecs are also of relevance to us. However, I think the speech codecs would be a good starting point. Steve

3 3

Looking for a solid C/C++ freeswitch developer
by Erik Lagerway 13 Jun '07

13 Jun '07

Gaboogie is dragging the conference calling and mobile conference calling industry out of the dark ages and into the bright, shiny world of Web 2.0. We're looking for talented, hard working and resourceful team members, who understand and appreciate life in an exciting start-up. LEAD GABOOGIE FREESWITCH DEVELOPER http://blog.gaboogie.com/2007/06/lead_gaboogie_freeswitch_devel.html If you are interested you can always Gaboogie me at the number below. Regards, -- Erik Lagerway Co-Founder Gaboogie Ph +1.604.562.VOIP (8647) erik(a)gaboogie.com http://www.gaboogie.com http://blog.gaboogie.com Gaboogie. Start On Time.

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ClueCon Announcement
by Anthony Minessale 13 Jun '07

13 Jun '07

Hi, I wanted to remind everyone that ClueCon is in a few weeks and if you plan to attend you should reserve your place now. Sangoma will be giving away a few FREE T1 and analog cards to some lucky atendees. Zap Micro will be giving a FREE 4 port analog card with 1 FXO and 1 FXS module to all attendees. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&…

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Re: [Freeswitch-users] codec and vad and cng configuration
by Anthony Minessale 10 Jun '07

10 Jun '07

----- Original Message ---- From: Mark D. Anderson <mda(a)discerning.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, June 9, 2007 1:18:09 PM Subject: [Freeswitch-users] codec and vad and cng configuration It seems FS will simply go through the rtpmap offers in an incoming INVITE, in the incoming order, and pick the first one that is in the codec-prefs in my sofia.conf.xml (see end of this message for trace). But I was expecting that FS (or sofia underneath) would instead negotiate by checking codecs in my configured list, and pick the first matching one in *my* list, not the first matching one in *their* list. --------------------------------------------------------------------------------------------------------------- That is correct. This was done intentionally. Since you asked *sooo* nicely I added a new param to the sofia profile: <param name="inbound-codec-negotiation" value="greedy"/> This will do what you *expected*, (sorry for not meeting *all* your needs). --------------------------------------------------------------------------------------------------------------- I see there is a "inbound-late-negotiation" param, but I don't know how this would be used -- nor do I think i'd want to use it, if the priority can just be taken from my list. --------------------------------------------------------------------------------------------------------------- This option prevents the codec negotiation from happening until the media is required. You can let calls hit the dialplan and examine the SDP with regular expressions and set an absolute codec string. The point here is to make it possible to control the codec behaviour from the code. Using this technique you can now set the greedy behaviour as a channel variable in the dialplan. Since you delayed negotation you had the oppurtunity to set the variable before the codec negotiation took place. <action application="set" data="sip_codec_negotiation=greedy"> <action application="answer"/> --------------------------------------------------------------------------------------------------------------- Regarding VAD, I'm wondering why there is an "in" or "both" setting -- isn't VAD chosen by the sender? Also, it seems kind of limiting to have to enable VAD regardless of who is connecting (what UA) and regardless of what codec is in use. And I see no way to control CNG at all? --------------------------------------------------------------------------------------------------------------- We *always* advertise CNG and we *always* set the CNG mode in the RTP if someone we talk to supports it. If they don't support it we turn it off. When this takes place you will see a message in the DEBUG log about setting the CNG payload type. This means the RTP stack will send CNG frames whenever it notices an interval take place where no audio was sent. The VAD param is the way it is because we have not worked it into the SDP negotiation yet. The in out or both is if you want to use this nonstandard VAD on the inbound media stream, the outbound streams or both. A small TIP. Going forward please take the time to read your email before you send it to make sure your wording or tone does not HIGHLY OFFEND the developers you are asking for help. YOU are asking ME to add functionality YOU want to MY software. I am perfectly happy without this behaviour and I am adding it for YOU. I'll give you the benefit of the doubt if you did not realize the demanding tone of your email but please pay more attention to your manners in the future. --------------------------------------------------------------------------------------------------------------- -mda example trace, showing how it picks PCMU since it is first in their list, when the first match in my list would have been iLBC ---- 2007-06-09 13:03:43 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 12720 12720 IN IP4 67.55.159.156 s=session c=IN IP4 67.55.159.156 t=0 0 m=audio 12104 RTP/AVP 0 8 3 111 5 10 7 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[speex:98:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[iLBC:97:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[GSM:3:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMA:8:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:817 sofia_glue_tech_set_codec() Set Codec sofia/profile1/8572337508@67.55.159.156:5060 PCMU/8000 20 ms _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/

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codec and vad and cng configuration
by Mark D. Anderson 09 Jun '07

09 Jun '07

It seems FS will simply go through the rtpmap offers in an incoming INVITE, in the incoming order, and pick the first one that is in the codec-prefs in my sofia.conf.xml (see end of this message for trace). But I was expecting that FS (or sofia underneath) would instead negotiate by checking codecs in my configured list, and pick the first matching one in *my* list, not the first matching one in *their* list. I see there is a "inbound-late-negotiation" param, but I don't know how this would be used -- nor do I think i'd want to use it, if the priority can just be taken from my list. Regarding VAD, I'm wondering why there is an "in" or "both" setting -- isn't VAD chosen by the sender? Also, it seems kind of limiting to have to enable VAD regardless of who is connecting (what UA) and regardless of what codec is in use. And I see no way to control CNG at all? -mda example trace, showing how it picks PCMU since it is first in their list, when the first match in my list would have been iLBC ---- 2007-06-09 13:03:43 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 12720 12720 IN IP4 67.55.159.156 s=session c=IN IP4 67.55.159.156 t=0 0 m=audio 12104 RTP/AVP 0 8 3 111 5 10 7 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[speex:98:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[iLBC:97:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[GSM:3:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMA:8:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:1157 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2007-06-09 13:03:43 [DEBUG] sofia_glue.c:817 sofia_glue_tech_set_codec() Set Codec sofia/profile1/8572337508@67.55.159.156:5060 PCMU/8000 20 ms

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Re: [Freeswitch-users] Authenticating calls
by Anthony Minessale 08 Jun '07

08 Jun '07

Ask and you shall receive.  <extension name="9191">  <condition field="destination_number" expression="^9191$"/>  <condition field="${sip_authorized}" expression="true"> <anti-action application="reject" data="407"/> </condition>  <condition> <action application="playback" data="/tmp/itworked.wav"/> </condition> </extension> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, June 8, 2007 12:25:36 PM Subject: Re: [Freeswitch-users] Authenticating calls I still also like the idea of an auth app that forces the call to be authenticated. Or maybe it could be a parameter that is passed to the sophia channel. This would be more secure then just checking if the user is allready registered. On 6/8/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > The best way to mix auth and not auth is to use > 2 profiles so you can make one be public and one private. > Since the private one is only for people to register it's > not inconvenient at all to make them use 5061 instead as it's > a 1 time setting. Otherwise you can just use 2 ip. > This way you can turn on full auth for the 1 profile and the other can be > directed to a paticular public context with a limited dialplan. > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Jason Garland <jgarland(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Friday, June 8, 2007 10:06:25 AM > Subject: Re: [Freeswitch-users] Authenticating calls > > If there was just a channel variable that told me if the call was from > a registered user or not I think I could use that too. > > On 6/8/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > > > We don't have that but it would fairly easy to add > > a dialplan application that could trigger a challange > > but from there you would need to have an external hook > > ready for the user directory when the auth happened. > > > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > JABBER:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Jason Garland <jgarland(a)gmail.com> > > To: freeswitch-users(a)lists.freeswitch.org > > Sent: Friday, June 8, 2007 8:42:02 AM > > Subject: Re: [Freeswitch-users] Authenticating calls > > > > Yes. Or check a condition that will tell me if it's a registered user. > > > > On 6/8/07, Mike Murdock <mmurdock(a)coppercom.com> wrote: > > > > > > > > > > > > So you want to get an invite, look at the headers, and if it is a toll > > call, > > > authenticate the subscriber? > > > > > > > > > Michael Murdock > > > VP Operations & Development > > > Switchmaxx Product Line > > > 125 N. Market. Suite 1520 > > > Wichita KS. 67202 > > > (316) 932-2693 > > > > > > > > > -----Original Message----- > > > From: Jason Garland [mailto:jgarland@gmail.com] > > > Sent: Thursday, June 07, 2007 10:58 PM Eastern Standard Time > > > To: freeswitch-users(a)lists.freeswitch.org > > > Subject: [Freeswitch-users] Authenticating > calls > > > > > > Is there something I can put in the dialplan to authenticate some > > > calls(Toll calls), and not others? > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ________________________________ > > Be a better Heartthrob. Get better relationship answers from someone who > > knows. > > Yahoo! Answers - Check it out. > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > The fish are biting. > Get more visitors on your site using Yahoo! Search Marketing. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. http://mobile.yahoo.com/go?refer=1GNXIC

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Re: [Freeswitch-users] Authenticating calls
by Anthony Minessale 08 Jun '07

08 Jun '07

The best way to mix auth and not auth is to use 2 profiles so you can make one be public and one private. Since the private one is only for people to register it's not inconvenient at all to make them use 5061 instead as it's a 1 time setting. Otherwise you can just use 2 ip. This way you can turn on full auth for the 1 profile and the other can be directed to a paticular public context with a limited dialplan. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, June 8, 2007 10:06:25 AM Subject: Re: [Freeswitch-users] Authenticating calls If there was just a channel variable that told me if the call was from a registered user or not I think I could use that too. On 6/8/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > We don't have that but it would fairly easy to add > a dialplan application that could trigger a challange > but from there you would need to have an external hook > ready for the user directory when the auth happened. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Jason Garland <jgarland(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Friday, June 8, 2007 8:42:02 AM > Subject: Re: [Freeswitch-users] Authenticating calls > > Yes. Or check a condition that will tell me if it's a registered user. > > On 6/8/07, Mike Murdock <mmurdock(a)coppercom.com> wrote: > > > > > > > > So you want to get an invite, look at the headers, and if it is a toll > call, > > authenticate the subscriber? > > > > > > Michael Murdock > > VP Operations & Development > > Switchmaxx Product Line > > 125 N. Market. Suite 1520 > > Wichita KS. 67202 > > (316) 932-2693 > > > > > > -----Original Message----- > > From: Jason Garland [mailto:jgarland@gmail.com] > > Sent: Thursday, June 07, 2007 10:58 PM Eastern Standard Time > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: [Freeswitch-users] Authenticating calls > > > > Is there something I can put in the dialplan to authenticate some > > calls(Toll calls), and not others? > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > Be a better Heartthrob. Get better relationship answers from someone who > knows. > Yahoo! Answers - Check it out. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

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Re: [Freeswitch-users] Authenticating calls
by Anthony Minessale 08 Jun '07

08 Jun '07

If you look at the last commit. I forgot to document the <variables> tag in the users. You can use that to set variables you want to exist when the call starts. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, June 8, 2007 10:06:25 AM Subject: Re: [Freeswitch-users] Authenticating calls If there was just a channel variable that told me if the call was from a registered user or not I think I could use that too. On 6/8/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > We don't have that but it would fairly easy to add > a dialplan application that could trigger a challange > but from there you would need to have an external hook > ready for the user directory when the auth happened. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Jason Garland <jgarland(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Friday, June 8, 2007 8:42:02 AM > Subject: Re: [Freeswitch-users] Authenticating calls > > Yes. Or check a condition that will tell me if it's a registered user. > > On 6/8/07, Mike Murdock <mmurdock(a)coppercom.com> wrote: > > > > > > > > So you want to get an invite, look at the headers, and if it is a toll > call, > > authenticate the subscriber? > > > > > > Michael Murdock > > VP Operations & Development > > Switchmaxx Product Line > > 125 N. Market. Suite 1520 > > Wichita KS. 67202 > > (316) 932-2693 > > > > > > -----Original Message----- > > From: Jason Garland [mailto:jgarland@gmail.com] > > Sent: Thursday, June 07, 2007 10:58 PM Eastern Standard Time > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: [Freeswitch-users] Authenticating calls > > > > Is there something I can put in the dialplan to authenticate some > > calls(Toll calls), and not others? > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > Be a better Heartthrob. Get better relationship answers from someone who > knows. > Yahoo! Answers - Check it out. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php

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Re: [Freeswitch-users] Authenticating calls
by Anthony Minessale 08 Jun '07

08 Jun '07

We don't have that but it would fairly easy to add a dialplan application that could trigger a challange but from there you would need to have an external hook ready for the user directory when the auth happened. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, June 8, 2007 8:42:02 AM Subject: Re: [Freeswitch-users] Authenticating calls Yes. Or check a condition that will tell me if it's a registered user. On 6/8/07, Mike Murdock <mmurdock(a)coppercom.com> wrote: > > > > So you want to get an invite, look at the headers, and if it is a toll call, > authenticate the subscriber? > > > Michael Murdock > VP Operations & Development > Switchmaxx Product Line > 125 N. Market. Suite 1520 > Wichita KS. 67202 > (316) 932-2693 > > > -----Original Message----- > From: Jason Garland [mailto:jgarland@gmail.com] > Sent: Thursday, June 07, 2007 10:58 PM Eastern Standard Time > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Authenticating calls > > Is there something I can put in the dialplan to authenticate some > calls(Toll calls), and not others? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ 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 ___________________________________________________________________________________ You snooze, you lose. Get messages ASAP with AutoCheck in the all-new Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_html.html

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